project('gstreamer-full', 'c',
- version : '1.19.3.1',
+ version : '1.19.90',
meson_version : '>= 0.59',
default_options : ['buildtype=debugoptimized',
# Needed due to https://github.com/mesonbuild/meson/issues/1889,
+=== release 1.19.90 ===
+
+2022-01-28 14:28:35 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * NEWS:
+ * RELEASE:
+ * gst-devtools.doap:
+ * meson.build:
+ Release 1.19.90
+
+2022-01-28 14:28:28 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * ChangeLog:
+ Update ChangeLogs for 1.19.90
+
2022-01-27 11:25:00 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
* meson_options.txt:
GStreamer 1.20 Release Notes
GStreamer 1.20 has not been released yet. It is scheduled for release in
-November 2021.
+late January / early February 2022.
1.19.x is the unstable development version that is being developed in
-the git main branch and which will eventually result in 1.20, and 1.19.3
-is the current development release in that series
-
-It is expected that feature freeze will be in early October 2021,
-followed by one or two 1.19.9x pre-releases and the new 1.20 stable
-release around October/November 2021.
+the git main branch and which will eventually result in 1.20, and
+1.19.90 is the first release candidate in that series (1.20rc1).
1.20 will be backwards-compatible to the stable 1.18, 1.16, 1.14, 1.12,
-1.10, 1.8, 1.6,, 1.4, 1.2 and 1.0 release series.
+1.10, 1.8, 1.6, 1.4, 1.2 and 1.0 release series.
See https://gstreamer.freedesktop.org/releases/1.20/ for the latest
version of this document.
-Last updated: Monday 1 November 2021, 01:00 UTC (log)
+Last updated: Wednesday 26 January 2022, 01:00 UTC (log)
Introduction
Highlights
-- this section will be completed in due course
+- Development in GitLab was switched to a single git repository
+ containing all the modules
+- GstPlay: new high-level playback library, replaces GstPlayer
+- WebM Alpha decoding support
+- Encoding profiles can now be tweaked with additional
+ application-specified element properties
+- Compositor: multi-threaded video conversion and mixing
+- RTP header extensions: unified support in RTP depayloader and
+ payloader base classes
+- SMPTE 2022-1 2-D Forward Error Correction support
+- Smart encoding (passthrough) support for VP8, VP9, H.265 in
+ encodebin and transcodebin
+- Runtime compatibility support for libsoup2 and libsoup3 (libsoup3
+ support experimental)
+- Video decoder subframe support
+- Video decoder automatic packet-loss, data corruption, and keyframe
+ request handling for RTP / WebRTC / RTSP
+- MP4 and Matroska muxers now support profile/level/resolution changes
+ for H264/H265 input streams (i.e. codec data changing on the fly)
+- MP4 muxing mode that initially creates a fragmented mp4 which is
+ converted to a regular mp4 on EOS
+- Audio support for the WebKit Port for Embedded (WPE) web page source
+ element
+- CUDA based video color space convert and rescale elements and
+ upload/download elements
+- NVIDIA memory:NVMM support for OpenGL glupload and gldownload
+ elements
+- Many WebRTC improvements
+- The new VA-API plugin implemention fleshed out with more decoders
+ and new postproc elements
+- AppSink API to retrieve events in addition to buffers and buffer
+ lists
+- AppSrc gained more configuration options for the internal queue
+ (leakiness, limits in buffers and time, getters to read current
+ levels)
+- Updated Rust bindings and many new Rust plugins
+- Improved support for custom minimal GStreamer builds
+- Support build against FFmpeg 5.0
+- Linux Stateless CODEC support gained MPEG2 and VP9
+- Windows Direct3D11/DXVA decoder gained AV1 and MPEG2 support
+- Lots of new plugins, features, performance improvements and bug
+ fixes
Major new features and changes
Noteworthy new features and API
-- this section will be filled in in due course
+- gst_element_get_request_pad() has been deprecated in favour of the
+ newly-added gst_element_request_pad_simple() which does the exact
+ same thing but has a less confusing name that hopefully makes clear
+ that the function request a new pad rather than just retrieves an
+ already-existing request pad.
+
+Development in GitLab was switched to a single git repository containing all the modules
+
+The GStreamer multimedia framework is a set of libraries and plugins
+split into a number of distinct modules which are released independently
+and which have so far been developed in separate git repositories in
+freedesktop.org GitLab.
+
+In addition to these separate git repositories there was a gst-build
+module that would use the Meson build systems’s subproject feature to
+download each individual module and then build everything in one go. It
+would also provide an uninstalled development environment that made it
+easy to work on GStreamer and use or test versions other than the
+system-installed GStreamer version.
+
+All of these modules have now (as of 28 September 2021) been merged into
+a single git repository (“Mono repository” or “monorepo”) which should
+simplify development workflows and continuous integration, especially
+where changes need to be made to multiple modules at once.
+
+This mono repository merge will primarily affect GStreamer developers
+and contributors and anyone who has workflows based on the GStreamer git
+repositories.
+
+The Rust bindings and Rust plugins modules have not been merged into the
+mono repository at this time because they follow a different release
+cycle.
+
+The mono repository lives in the existing GStreamer core git repository
+in GitLab in the new main branch and all future development will happen
+on this branch.
+
+Modules will continue to be released as separate tarballs.
+
+For more details, please see the GStreamer mono repository FAQ.
+
+GstPlay: new high-level playback library replacing GstPlayer
+
+- GstPlay is a new high-level playback library that replaces the older
+ GstPlayer API. It is basically the same API as GstPlayer but
+ refactored to use bus messages for application notifications instead
+ of GObject signals. There is still a signal adapter object for those
+ who prefer signals. Since the existing GstPlayer API is already in
+ use in various applications, it didn’t seem like a good idea to
+ break it entirely. Instead a new API was added, and it is expected
+ that this new GstPlay API will be moved to gst-plugins-base in
+ future.
+
+- The existing GstPlayer API is scheduled for deprecation and will be
+ removed at some point in the future (e.g. in GStreamer 1.24), so
+ application developers are urged to migrate to the new GstPlay API
+ at their earliest convenience.
+
+WebM alpha decoding
+
+- Implement WebM alpha decoding (VP8/VP9 with alpha), which required
+ support and additions in various places. This is supported both with
+ software decoders and hardware-accelerated decoders.
+
+- VP8/VP9 don’t support alpha components natively in the codec, so the
+ way this is implemented in WebM is by encoding the alpha plane with
+ transparency data as a separate VP8/VP9 stream. Inside the WebM
+ container (a variant of Matroska) this is coded as a single video
+ track with the “normal” VP8/VP9 video data making up the main video
+ data and each frame of video having an encoded alpha frame attached
+ to it as extra data ("BlockAdditional").
+
+- matroskademux has been extended extract this per-frame alpha side
+ data and attach it in form of a GstVideoCodecAlphaMeta to the
+ regular video buffers. Note that this new meta is specific to this
+ VP8/VP9 alpha support and can’t be used to just add alpha support to
+ other codecs that don’t support it. Lastly, matroskademux also
+ advertises the fact that the streams contain alpha in the caps.
+
+- The new codecalpha plugin contains various bits of infrastructure to
+ support autoplugging and debugging:
+
+ - codecalphademux splits out the alpha stream from the metas on
+ the regular VP8/VP9 buffers
+ - alphacombine takes two decoded raw video streams (one alpha, one
+ the regular video) and combines it into a video stream with
+ alpha
+ - vp8alphadecodebin + vp9alphadecodebin are wrapper bins that use
+ the regular vp8dec and vp9dec software decoders to decode
+ regular and alpha streams and combine them again. To decodebin
+ these look like regular decoders which ju
+ - The V4L2 CODEC plugin has stateless VP8/VP9 decoders that can
+ decode both alpha and non-alpha stream with a single decoder
+ instance
+
+- A new AV12 video format was added which is basically NV12 with an
+ alpha plane, which is more convenient for many hardware-accelerated
+ decoders.
+
+- Watch Nicolas Dufresne’s LCA 2022 talk “Bringing WebM Alpha support
+ to GStreamer” for all the details and a demo.
+
+RTP Header Extensions Base Class and Automatic Header Extension Handling in RTP Payloaders and Depayloaders
+
+- RTP Header Extensions are specified in RFC 5285 and provide a way to
+ add small pieces of data to RTP packets in between the RTP header
+ and the RTP payload. This is often used for per-frame metadata,
+ extended timestamps or other application-specific extra data. There
+ are several commonly-used extensions specified in various RFCs, but
+ senders are free to put any kind of data in there, as long as sender
+ and receiver both know what that data is. Receivers that don’t know
+ about the header extensions will just skip the extra data without
+ ever looking at it. These header extensions can often be combined
+ with any kind of payload format, so may need to be supported by many
+ RTP payloader and depayloader elements.
+
+- Inserting and extracting RTP header extension data has so far been a
+ bit inconvenient in GStreamer: There are functions to add and
+ retrieve RTP header extension data from RTP packets, but nothing
+ works automatically, even for common extensions. People would have
+ to do the insertion/extraction either in custom elements
+ before/after the RTP payloader/depayloader, or inside pad probes,
+ which isn’t very nice.
+
+- This release adds various pieces of new infrastructure for generic
+ RTP header extension handling, as well as some implementations for
+ common extensions:
+
+ - GstRTPHeaderExtension is a new helper base class for reading and
+ writing RTP header extensions. Nominally this subclasses
+ GstElement, but only so these extensions are stored in the
+ registry where they can be looked up by URI or name. They don’t
+ have pads and don’t get added to the pipeline graph as an
+ element.
+
+ - "add-extension" and "clear-extension" action signals on RTP
+ payloaders and depayloaders for manual extension management
+
+ - The "request-extension" signal will be emitted if an extension
+ is encountered that requires explicit mapping by the application
+
+ - new "auto-header-extension" property on RTP payloaders and
+ depayloaders for automatic handling of known header extensions.
+ This is enabled by default. The extensions must be signalled via
+ caps / SDP.
+
+ - RTP header extension implementations:
+
+ - rtphdrextclientaudiolevel: Client-to-Mixer Audio Level
+ Indication (RFC 6464) (also see below)
+ - rtphdrextcolorspace: Color Space extension, extends RTP
+ packets with color space and high dynamic range (HDR)
+ information
+ - rtphdrexttwcc: Transport Wide Congestion Control support
+
+- gst_rtp_buffer_remove_extension_data() is a new helper function to
+ remove an RTP header extension from an RTP buffer
+
+- The existing gst_rtp_buffer_set_extension_data() now also supports
+ shrinking the extension data in size
+
+AppSink and AppSrc improvements
+
+- appsink: new API to pull events out of appsink in addition to
+ buffers and buffer lists.
+
+ There was previously no way for users to receive incoming events
+ from appsink properly serialised with the data flow, even if they
+ are serialised events. The reason for that is that the only way to
+ intercept events was via a pad probe on the appsink sink pad, but
+ there is also internal queuing inside of appsink, so it’s difficult
+ to ascertain the right order of everything in all cases.
+
+ There is now a new "new-serialized-event" signal which will be
+ emitted when there’s a new event pending (just like the existing
+ "new-sample" signal). The "emit-signals" property must be set to
+ TRUE in order to activate this (but it’s also fine to just pull from
+ the application thread without using the signals).
+
+ gst_app_sink_pull_object() and gst_app_sink_try_pull_object() can be
+ used to pull out either an event or a new sample carrying a buffer
+ or buffer list, whatever is next in the queue.
+
+ EOS events will be filtered and will not be returned. EOS handling
+ can be done the ususal way, same as with _pull_sample().
+
+- appsrc: allow configuration of internal queue limits in time and
+ buffers and add leaky mode.
+
+ There is internal queuing inside appsrc so the application thread
+ can push data into the element which will then be picked up by the
+ source element’s streaming thread and pushed into the pipeline from
+ that streaming thread. This queue is unlimited by default and until
+ now it was only possible to set a maximum size limit in bytes. When
+ that byte limit is reached, the pushing thread (application thread)
+ would be blocked until more space becomes available.
+
+ A limit in bytes is not particularly useful for many use cases, so
+ now it is possible to also configure limits in time and buffers
+ using the new "max-time" and "max-buffers" properties. Of course
+ there are also matching new read-only"current-level-buffers" and
+ "current-level-time properties" properties to query the current fill
+ level of the internal queue in time and buffers.
+
+ And as if that wasn’t enough the internal queue can also be
+ configured as leaky using the new "leaky-type" property. That way
+ when the queue is full the application thread won’t be blocked when
+ it tries to push in more data, but instead either the new buffer
+ will be dropped or the oldest data in the queue will be dropped.
+
+Better string serialization of nested GstCaps and GstStructures
+
+- New string serialisation format for structs and caps that can handle
+ nested structs and caps properly by using brackets to delimit nested
+ items (e.g. some-struct, some-field=[nested-struct, nested=true]).
+ Unlike the default format the new variant can also support more than
+ one level of nesting. For backwards-compatibility reasons the old
+ format is still output by default when serialising caps and structs
+ using the existing API. The new functions gst_caps_serialize() and
+ gst_structure_serialize() can be used to output strings in the new
+ format.
+
+Convenience API for custom GstMetas
+
+- New convenience API to register and create custom GstMetas:
+ gst_meta_register_custom() and gst_buffer_add_custom_meta(). Such
+ custom meta is backed by a GstStructure and does not require that
+ users of the API expose their GstMeta implementation as public API
+ for other components to make use of it. In addition, it provides a
+ simpler interface by ignoring the impl vs. api distinction that the
+ regular API exposes. This new API is meant to be the meta
+ counterpart to custom events and messages, and to be more convenient
+ than the lower-level API when the absolute best performance isn’t a
+ requirement. The reason it’s less performant than a “proper” meta is
+ that a proper meta is just a C struct in the end whereas this goes
+ through the GstStructure API which has a bit more overhead, which
+ for most scenarios is negligible however. This new API is useful for
+ experimentation or proprietary metas, but also has some limitations:
+ it can only be used if there’s a single producer of these metas;
+ it’s not allowed to register the same custom meta multiple times or
+ from multiple places.
+
+Additional Element Properties on Encoding Profiles
+
+- GstEncodingProfile: The new "element-properties" and
+ gst_encoding_profile_set_element_properties() API allows
+ applications to set additional element properties on encoding
+ profiles to configure muxers and encoders. So far the encoding
+ profile template was the only place where this could be specified,
+ but often what applications want to do is take a ready-made encoding
+ profile shipped by GStreamer or the application and then tweak the
+ settings on top of that, which is now possible with this API. Since
+ applications can’t always know in advance what encoder element will
+ be used in the end, it’s even possible to specify properties on a
+ per-element basis.
+
+ Encoding Profiles are used in the encodebin, transcodebin and
+ camerabin elements and APIs to configure output formats (containers
+ and elementary streams).
+
+Audio Level Indication Meta for RFC 6464
+
+- New GstAudioLevelMeta containing Audio Level Indication as per RFC
+ 6464
+
+- The level element has been updated to add GstAudioLevelMeta on
+ buffers if the "audio-level-meta" property is set to TRUE. This can
+ then in turn be picked up by RTP payloaders to signal the audio
+ level to receivers through RTP header extensions (see above).
+
+- New Client-to-Mixer Audio Level Indication (RFC6464) RTP Header
+ Extension which should be automatically created and used by RTP
+ payloaders and depayloaders if their "auto-header-extension"
+ property is enabled and if the extension is part of the RTP caps.
+
+Automatic packet loss, data corruption and keyframe request handling for video decoders
+
+- The GstVideoDecoder base class has gained various new APIs to
+ automatically handle packet loss and data corruption better by
+ default, especially in RTP, RTSP and WebRTC streaming scenarios, and
+ to give subclasses more control about how they want to handle
+ missing data:
+
+ - Video decoder subclasses can mark output frames as corrupted via
+ the new GST_VIDEO_CODEC_FRAME_FLAG_CORRUPTED flag
+
+ - A new "discard-corrupted-frames" property allows applications to
+ configure decoders so that corrupted frames are directly
+ discarded instead of being forwarded inside the pipeline. This
+ is a replacement for the "output-corrupt" property of the FFmpeg
+ decoders.
+
+ - RTP depayloaders can now signal to decoders that data is missing
+ when sending GAP events for lost packets. GAP events can be sent
+ for various reason in a GStreamer pipeline. Often they are just
+ used to let downstream elements know that there isn’t a buffer
+ available at the moment, so downstream elements can move on
+ instead of waiting for one. They are also sent by RTP
+ depayloaders in the case that packets are missing, however, and
+ so far a decoder was not able to differentiate the two cases.
+ This has been remedied now: GAP events can be decorated with
+ gst_event_set_gap_flags() and GST_GAP_FLAG_MISSING_DATA to let
+ decoders now what happened, and decoders can then use that in
+ some cases to handle missing data better.
+
+ - The GstVideoDecoder::handle_missing_data vfunc was added to
+ inform subclasses about packet loss or missing data and let them
+ handle it in their own way if they like.
+
+ - gst_video_decoder_set_needs_sync_point() lets subclasses signal
+ that they need the stream to start with a sync point. If
+ enabled, the base class will discard all non-sync point frames
+ in the beginning and after a flush and does not pass them to the
+ subclass. Furthermore, if the first frame is not a sync point,
+ the base class will try and request a sync frame from upstream
+ by sending a force-key-unit event (see next items).
+
+ - New "automatic-request-sync-points" and
+ "automatic-request-sync-point-flags" properties to automatically
+ request sync points when needed, e.g. on packet loss or if the
+ first frame is not a keyframe. Applications may want to enable
+ this on decoders operating in e.g. RTP/WebRTC/RTSP receiver
+ pipelines.
+
+ - The new "min-force-key-unit-interval" property can be used to
+ ensure there’s a minimal interval between keyframe requests to
+ upstream (and/or the sender) and we’re not flooding the sender
+ with key unit requests.
+
+ - gst_video_decoder_request_sync_point() allows subclasses to
+ request a new sync point (e.g. if they choose to do their own
+ missing data handling). This will still honour the
+ "min-force-key-unit-interval" property if set.
+
+Improved support for custom minimal GStreamer builds
+
+- Element registration and registration of other plugin features
+ inside plugin init functions has been improved in order to
+ facilitate minimal custom GStreamer builds.
+
+- A number of new macros have been added to declare and create
+ per-element and per-pluginfeature register functions in all plugins,
+ and then call those from the per-plugin plugin_init functions:
+
+ - GST_ELEMENT_REGISTER_DEFINE,
+ GST_DEVICE_PROVIDER_REGISTER_DEFINE,
+ GST_DYNAMIC_TYPE_REGISTER_DEFINE, GST_TYPE_FIND_REGISTER_DEFINE
+ for the actual registration call with GStreamer
+ - GST_ELEMENT_REGISTER, GST_DEVICE_PROVIDER_REGISTER,
+ GST_DYNAMIC_TYPE_REGISTER, GST_PLUGIN_STATIC_REGISTER,
+ GST_TYPE_FIND_REGISTER to call the registration function defined
+ by the REGISTER_DEFINE macro
+ - GST_ELEMENT_REGISTER_DECLARE,
+ GST_DEVICE_PROVIDER_REGISTER_DECLARE,
+ GST_DYNAMIC_TYPE_REGISTER_DECLARE,
+ GST_TYPE_FIND_REGISTER_DECLARE to declare the registration
+ function defined by the REGISTER_DEFINE macro
+ - and various variants for advanced use cases.
+
+- This means that applications can call the per-element and
+ per-pluginfeature registration functions for only the elements they
+ need instead of registering plugins as a whole with all kinds of
+ elements that may not be required (e.g. encoder and decoder instead
+ of just decoder). In case of static linking all unused functions and
+ their dependencies would be removed in this case by the linker,
+ which helps minimise binary size for custom builds.
+
+- gst_init() will automatically call a gst_init_static_plugins()
+ function if one exists.
+
+- See the GStreamer static build documentation and Stéphane’s blog
+ post Generate a minimal GStreamer build, tailored to your needs for
+ more details.
New elements
-- this section will be filled in in due course
+- New aesdec and aesenc elements for AES encryption and decryption in
+ a custom format.
+
+- New encodebin2 element with dynamic/sometimes source pads in order
+ to support the option of doing the muxing outside of encodebin,
+ e.g. in combination with a splitmuxsink.
+
+- New fakeaudiosink and videocodectestsink elements for testing and
+ debugging (see below for more details)
+
+- rtpisacpay, rtpisacdepay: new RTP payloader and depayloader for iSAC
+ audio codec
+
+- rtpst2022-1-fecdec, rtpst2022-1-fecenc: new elements providing SMPTE
+ 2022-1 2-D Forward Error Correction. More details in Mathieu’s blog
+ post.
+
+- isac: new plugin wrapping the Internet Speech Audio Codec reference
+ encoder and decoder from the WebRTC project.
+
+- asio: plugin for Steinberg ASIO (Audio Streaming Input/Output) API
+
+- gssrc, gssink: add source and sink for Google Cloud Storage
+
+- onnx: new plugin to apply ONNX neural network models to video
+
+- openaptx: aptX and aptX-HD codecs using libopenaptx (v0.2.0)
+
+- qroverlay, debugqroverlay: new elements that allows overlaying data
+ on top of video in form of a QR code
+
+- cvtracker: new OpenCV-based tracker element
+
+- av1parse, vp9parse: new parsers for AV1 and VP9 video
+
+- va: work on the new VA-API plugin implementation for
+ hardware-accelerated video decoding and encoding has continued at
+ pace, with various new decoders and filters having joined the
+ initial vah264dec:
+
+ - vah265dec: VA-API H.265 decoder
+ - vavp8dec: VA-API VP8 decoder
+ - vavp9dec: VA-API VP9 decoder
+ - vaav1dec: VA-API AV1 decoder
+ - vampeg2dec: VA-API MPEG-2 decoder
+ - vadeinterlace: : VA-API deinterlace filter
+ - vapostproc: : VA-API postproc filter (color conversion,
+ resizing, cropping, color balance, video rotation, skin tone
+ enhancement, denoise, sharpen)
+
+ See Víctor’s blog post “GstVA in GStreamer 1.20” for more details
+ and what’s coming up next.
+
+- vaapiav1dec: new AV1 decoder element (in gstreamer-vaapi)
+
+- msdkav1dec: hardware-accelerated AV1 decoder using the Intel Media
+ SDK / oneVPL
+
+- nvcodec plugin for NVIDIA NVCODEC API for hardware-accelerated video
+ encoding and decoding:
+
+ - cudaconvert, cudascale: new CUDA based video color space convert
+ and rescale elements
+ - cudaupload, cudadownload: new helper elements for memory
+ transfer between CUDA and system memory spaces
+ - nvvp8sldec, nvvp9sldec: new GstCodecs-based VP8/VP9 decoders
+
+- Various new hardware-accelerated elements for Windows:
+
+ - d3d11screencapturesrc: new desktop capture element, including a
+ GstDeviceProvider implementation to enumerate/select target
+ monitors for capture.
+ - d3d11av1dec and d3d11mpeg2dec: AV1 and MPEG-2 decoders
+ - d3d11deinterlace: deinterlacing filter
+ - d3d11compositor: video composing element
+ - see Windows section below for more details
+
+- new Rust plugins:
+
+ - audiornnoise: Removes noise from an audio stream
+ - awstranscribeparse: Parses AWS audio transcripts into timed text
+ buffers
+ - ccdetect: Detects if valid closed captions are present in a
+ closed captions stream
+ - cea608tojson: Converts CEA-608 Closed Captions to a JSON
+ representation
+ - cmafmux: CMAF fragmented MP4 muxer
+ - dashmp4mux: DASH fragmented MP4 muxer
+ - isofmp4mux: ISO fragmented MP4 muxer
+ - ebur128level: EBU R128 Loudness Level Measurement
+ - ffv1dec: FFV1 video decoder
+ - gtk4paintablesink: GTK4 video sink, which provides a
+ GdkPaintable that can be rendered in various widgets
+ - hlssink3: HTTP Live Streaming sink
+ - hrtfrender: Head-Related Transfer Function (HRTF) renderer
+ - hsvdetector: HSV colorspace detector
+ - hsvfilter: HSV colorspace filter
+ - jsongstenc: Wraps buffers containing any valid top-level JSON
+ structures into higher level JSON objects, and outputs those as
+ ndjson
+ - jsongstparse: Parses ndjson as output by jsongstenc
+ - jsontovtt: converts JSON to WebVTT subtitles
+ - regex: Applies regular expression operations on text
+ - roundedcorners: Adds rounded corners to video
+ - spotifyaudiosrc: Spotify source
+ - textahead: Display upcoming text buffers ahead (e.g. for
+ Karaoke)
+ - transcriberbin: passthrough bin that transcribes raw audio to
+ closed captions using awstranscriber and puts the captions as
+ metas onto the video
+ - tttojson: Converts timed text to a JSON representation
+ - uriplaylistbin: Playlist source bin
+ - webpdec-rs: WebP image decoder with animation support
+
+- New plugin codecalpha with elements to assist with WebM Alpha
+ decoding
+
+ - codecalphademux: Split stream with GstVideoCodecAlphaMeta into
+ two streams
+ - alphacombine: Combine two raw video stream (I420 or NV12) as one
+ stream with alpha channel (A420 or AV12)
+ - vp8alphadecodebin: A bin to handle software decoding of VP8 with
+ alpha
+ - vp9alphadecodebin: A bin to handle software decoding of VP9 with
+ alpha
+
+- New hardware accelerated elements for Linux:
+
+ - v4l2slmpeg2dec: Support for Linux Stateless MPEG2 decoders
+ - v4l2slvp9dec: Support for Linux Stateless VP9 decoders
+ - v4l2slvp8alphadecodebin: Support HW accelerated VP8 with alpha
+ layer decoding
+ - v4l2slvp9alphadecodebin: Support HW accelerated VP9 with alpha
+ layer decoding
New element features and additions
-- this section will be filled in in due course
+- assrender: handle more font mime types; better interaction with
+ matroskademux for embedded fonts
+
+- audiobuffersplit: Add support for specifying output buffer size in
+ bytes (not just duration)
+
+- audiolatency: new "samplesperbuffer" property so users can configure
+ the number of samples per buffer. The default value is 240 samples
+ which is equivalent to 5ms latency with a sample rate of 48000,
+ which might be larger than actual buffer size of audio capture
+ device.
+
+- audiomixer, audiointerleave, GstAudioAggregator: now keep a count of
+ samples that are dropped or processed as statistic and can be made
+ to post QoS messages on the bus whenever samples are dropped by
+ setting the "qos-messages" property on input pads.
+
+- audiomixer, compositor: improved handling of new inputs added at
+ runtime. New API was added to the GstAggregator base class to allow
+ subclasses to opt into an aggregation mode where inactive pads are
+ ignored when processing input buffers
+ (gst_aggregator_set_ignore_inactive_pads(),
+ gst_aggregator_pad_is_inactive()). An “inactive pad” in this context
+ is a pad which, in live mode, hasn’t yet received a first buffer,
+ but has been waited on at least once. What would happen usually in
+ this case is that the aggregator would wait for data on this pad
+ every time, up to the maximum configured latency. This would
+ inadvertently push mixer elements in live mode to the configured
+ latency envelope and delay processing when new inputs are added at
+ runtime until these inputs have actually produced data. This is
+ usually undesirable. With this new API, new inputs can be added
+ (requested) and configured and they won’t delay the data processing.
+ Applications can opt into this new behaviour by setting the
+ "ignore-inactive-pads" property on compositor, audiomixer or other
+ GstAudioAggregator-based elements.
+
+- cccombiner: implement “scheduling” of captions. So far cccombiner’s
+ behaviour was essentially that of a funnel: it strictly looked at
+ input timestamps to associate together video and caption buffers.
+ Now it will try to smoothly schedule caption buffers in order to
+ have exactly one per output video buffer. This might involve
+ rewriting input captions, for example when the input is CDP then
+ sequence counters are rewritten, time codes are dropped and
+ potentially re-injected if the input video frame had a time code
+ meta. This can also lead to the input drifting from synchronization,
+ when there isn’t enough padding in the input stream to catch up. In
+ that case the element will start dropping old caption buffers once
+ the number of buffers in its internal queue reaches a certain limit
+ (configurable via the "max-scheduled" property). The new original
+ funnel-like behaviour can be restored by setting the "scheduling"
+ property to FALSE.
+
+- ccconverter: new "cdp-mode" property to specify which sections to
+ include in CDP packets (timecode, CC data, service info). Various
+ software, including ffmpeg’s Decklink support, fails parsing CDP
+ packets that contain anything but CC data in the CDP packets.
+
+- clocksync: new "sync-to-first" property for automatic timestamp
+ offset setup: if set clocksync will set up the "ts-offset" value
+ based on the first buffer and the pipeline’s running time when the
+ first buffer arrived. The newly configured "ts-offset" in this case
+ would be the value that allows outputting the first buffer without
+ waiting on the clock. This is useful for example to feed a non-live
+ input into an already-running pipeline.
+
+- compositor:
+
+ - multi-threaded input conversion and compositing. Set the
+ "max-threads" property to activate this.
+ - new "sizing-policy" property to support display aspect ratio
+ (DAR)-aware scaling. By default the image is scaled to fill the
+ configured destination rectangle without padding and without
+ keeping the aspect ratio. With sizing-policy=keep-aspect-ratio
+ the input image is scaled to fit the destination rectangle
+ specified by GstCompositorPad:{xpos, ypos, width, height}
+ properties preserving the aspect ratio. As a result, the image
+ will be centered in the destination rectangle with padding if
+ necessary.
+ - new "zero-size-is-unscaled" property on input pads. By default
+ pad width=0 or pad height=0 mean that the stream should not be
+ scaled in that dimension. But if the "zero-size-is-unscaled"
+ property is set to FALSE a width or height of 0 is instead
+ interpreted to mean that the input image on that pad should not
+ be composited, which is useful when creating animations where an
+ input image is made smaller and smaller until it disappears.
+ - improved handling of new inputs at runtime via
+ "ignore-inactive-pads"property (see above for details)
+ - allow output format with alpha even if none of the inputs have
+ alpha (also glvideomixer and other GstVideoAggregator
+ subclasses)
+
+- dashsink: add h265 codec support and signals for allowing custom
+ playlist/fragment output
+
+- decodebin3:
+
+ - improved decoder selection, especially for hardware decoders
+ - make input activation “atomic” when adding inputs dynamically
+ - better interleave handling: take into account decoder latency
+ for interleave size
+
+- decklink:
+
+ - Updated DeckLink SDK to 11.2 to support DeckLink 8K Pro
+ - decklinkvideosrc:
+ - More accurate and stable capture timestamps: use the
+ hardware reference clock time when the frame was finished
+ being captured instead of a clock time much further down the
+ road.
+ - Automatically detect widescreen vs. normal NTSC/PAL
+
+- encodebin:
+
+ - add “smart encoding” support for H.265, VP8 and VP9 (i.e. only
+ re-encode where needed and otherwise pass through encoded video
+ as-is).
+ - H264/H265 smart encoding improvements: respect user-specified
+ stream-format, but if not specified default to avc3/hvc1 with
+ in-band SPS/PPS/VPS signalling for more flexibility.
+ - new encodebin2 element with dynamic/sometimes source pads in
+ order to support the option of doing the muxing outside of
+ encodebin, e.g. in combination with splitmuxsink.
+ - add APIs to set element properties on encoding profiles (see
+ below)
+
+- errorignore: new "ignore-eos" property to also ignore FLOW_EOS from
+ downstream elements
+
+- giosrc: add support for growing source files: applications can
+ specify that the underlying file being read is growing by setting
+ the "is-growing" property. If set, the source won’t EOS when it
+ reaches the end of the file, but will instead start monitoring it
+ and will start reading data again whenever a change is detected. The
+ new "waiting-data" and "done-waiting-data" signals keep the
+ application informed about the current state.
+
+- gtksink, gtkglsink:
+
+ - scroll event support: forwarded as navigation events into the
+ pipeline
+ - "video-aspect-ratio-override" property to force a specific
+ aspect ratio
+ - "rotate-method" property and support automatic rotation based on
+ image tags
+
+- identity: new "stats" property allows applications to retrieve the
+ number of bytes and buffers that have passed through so far.
+
+- interlace: add support for more formats, esp 10-bit, 12-bit and
+ 16-bit ones
+
+- jack: new "low-latency" property for automatic latency-optimized
+ setting and "port-names" property to select ports explicitly
+
+- jpegdec: support output conversion to RGB using libjpeg-turbo (for
+ certain input files)
+
+- line21dec:
+
+ - "mode" property to control whether and how detected closed
+ captions should be inserted in the list of existing close
+ caption metas on the input frame (if any): add, drop, or
+ replace.
+ - "ntsc-only" property to only look for captions if video has NTSC
+ resolution
+
+- line21enc: new "remove-caption-meta" to remove metas from output
+ buffers after encoding the captions into the video data; support for
+ CDP closed captions
+
+- matroskademux, matroskamux: Add support for ffv1, a lossless
+ intra-frame video coding format.
+
+- matroskamux: accept in-band SPS/PPS/VPS for H264 and H265
+ (i.e. stream-format avc3 and hev1) which allows on-the-fly
+ profile/level/resolution changes.
+
+- matroskamux: new "cluster-timestamp-offset" property, useful for use
+ cases where the container timestamps should map to some absolute
+ wall clock time, for example.
+
+- rtpsrc: add "caps" property to allow explicit setting of the caps
+ where needed
+
+- mpegts: support SCTE-35 passthrough via new "send-scte35-events"
+ property on MPEG-TS demuxer tsdemux. When enabled, SCTE 35 sections
+ (eg ad placement opportunities) are forwarded as events donwstream
+ where they can be picked up again by mpegtsmux. This required a
+ semantic change in the SCTE-35 section API: timestamps are now in
+ running time instead of muxer pts.
+
+- tsdemux: Handle PCR-less MPEG-TS streams; more robust timestamp
+ handling in certain corner cases and for poorly muxed streams.
+
+- mpegtsmux:
+
+ - More conformance improvements to make MPEG-TS analyzers happy:
+ - PCR timing accuracy: Improvements to the way mpegtsmux
+ outputs PCR observations in CBR mode, so that a PCR
+ observation is always inserted when needed, so that we never
+ miss the configured pcr-interval, as that triggers various
+ MPEG-TS analyser errors.
+ - Improved PCR/SI scheduling
+ - Don’t write PCR until PAT/PMT are output to make sure streams
+ start cleanly with a PAT/PMT.
+ - Allow overriding the automatic PMT PID selection via
+ application-supplied PMT_%d fields in the prog-map
+ structure/property.
+
+- mp4mux:
+
+ - new "first-moov-then-finalise" mode for fragmented output where
+ the output will start with a self-contained moov atom for the
+ first fragment, and then produce regular fragments. Then at the
+ end when the file is finalised, the initial moov is invalidated
+ and a new moov is written covering the entire file. This way the
+ file is a “fragmented mp4” file while it is still being written
+ out, and remains playable at all times, but at the end it is
+ turned into a regular mp4 file (with former fragment headers
+ remaining as unused junk data in the file).
+ - support H.264 avc3 and H.265 hvc1 stream formats as input where
+ the codec data is signalled in-band inside the bitstream instead
+ of caps/file headers.
+ - support profile/level/resolution changes for H264/H265 input
+ streams (i.e. codec data changing on the fly). Each codec_data
+ is put into its own SampleTableEntry inside the stsd, unless the
+ input is in avc3 stream format in which case it’s written
+ in-band an not in the headers.
+
+- multifilesink: new ""min-keyframe-distance"" property to make
+ minimum distance between keyframes in next-file=key-frame mode
+ configurable instead of hard-coding it to 10 seconds.
+
+- mxfdemux has seen a big refactoring to support non-frame wrappings
+ and more accurate timestamp/seek handling for some formats
+
+- msdk plugin for hardware-accelerated video encoding and decoding
+ using the Intel Media SDK:
+
+ - oneVPL support (Intel oneAPI Video Processing Library)
+ - AV1 decoding support
+ - H.264 decoder now supports constrained-high and progressive-high
+ profiles
+ - H.264 encoder:
+ - more configuration options (properties):
+ "intra-refresh-type", "min-qp" , "max-qp", "p-pyramid",
+ "dblk-idc"
+ - H.265 encoder:
+ - can output main-still-picture profile
+ - now inserts HDR SEIs (mastering display colour volume and
+ content light level)
+ - more configuration options (properties):
+ "intra-refresh-type", "min-qp" , "max-qp", "p-pyramid",
+ "b-pyramid", "dblk-idc", "transform-skip"
+ - support for RGB 10bit format
+ - External bitrate control in encoders
+ - Video post proc element msdkvpp gained support for 12-bit pixel
+ formats P012_LE, Y212_LE and Y412_LE
+
+- nvh264sldec: interlaced stream support
+
+- openh264enc: support main, high, constrained-high and
+ progressive-high profiles
+
+- openjpeg: support for multithreaded decoding and encoding
+
+- rtspsrc: now supports IPv6 also for tunneled mode (RTSP-over-HTTP);
+ new "ignore-x-server-reply" property to ignore the
+ x-server-ip-address server header reply in case of HTTP tunneling,
+ as it is often broken.
+
+- souphttpsrc: Runtime compatibility support for libsoup2 and
+ libsoup3. libsoup3 is the latest major version of libsoup, but
+ libsoup2 and libsoup3 can’t co-exist in the same process because
+ there is no namespacing or versioning for GObject types. As a
+ result, it would be awkward if the GStreamer souphttpsrc plugin
+ linked to a specific version of libsoup, because it would only work
+ with applications that use the same version of libsoup. To make this
+ work, the soup plugin now tries to determine the libsoup version
+ used by the application (and its other dependencies) at runtime on
+ systems where GStreamer is linked dynamically. libsoup3 support is
+ still considered somewhat experimental at this point.
+
+- srtsrc, srtsink: add signals for the application to accept/reject
+ incoming connections
+
+- timeoverlay: new elapsed-running-time time mode which shows the
+ running time since the first running time (and each flush-stop).
+
+- udpsrc: new timestamping mode to retrieve packet receive timestamps
+ from the kernel via socket control messages (SO_TIMESTAMPNS) on
+ supported platforms
+
+- uritranscodebin: new setup-source and element-setup signals for
+ applications to configure elements used
+
+- v4l2codecs plugin gained support for 4x4 and 32x32 tile formats
+ enabling some platforms or direct renders. Important memory usage
+ improvement.
+
+- v4l2slh264dec now implements the final Linux uAPI as shipped on
+ Linux 5.11 and later.
+
+- valve: add "drop-mode" property and provide two new modes of
+ operation: in drop-mode=forward-sticky-events sticky events
+ (stream-start, segment, tags, caps, etc.) are forwarded downstream
+ even when dropping is enabled; drop-mode=transform-to-gap will in
+ addition also convert buffers into gap events when dropping is
+ enabled, which lets downstream elements know that time is advancing
+ and might allow for preroll in many scenarios. By default all events
+ and all buffers are dropped when dropping is enabled, which can
+ cause problems with caps negotiation not progressing or branches not
+ prerolling when dropping is enabled.
+
+- videocrop: support for many more pixel formats, e.g. planar YUV
+ formats with > 8bits and GBR* video formats; can now also accept
+ video not backed by system memory as long as downstream supports the
+ GstCropMeta
+
+- videotestsrc: new smpte-rp-219 pattern for SMPTE75 RP-219 conformant
+ color bars
+
+- vp8enc: finish support for temporal scalability: two new properties
+ ("temporal-scalability-layer-flags",
+ "temporal-scalability-layer-sync-flags") and a unit change on the
+ "temporal-scalability-target-bitrate" property (now expects bps);
+ also make temporal scalability details available to RTP payloaders
+ as buffer metadata.
+
+- vp9enc: new properties to tweak encoder performance:
+
+ - "aq-mode" to configure adaptive quantization modes
+ - "frame-parallel-decoding" to configure whether to create a
+ bitstream that reduces decoding dependencies between frames
+ which allows staged parallel processing of more than one video
+ frames in the decoder. (Defaults to TRUE)
+ - "row-mt", "tile-columns" and "tile-rows" so multithreading can
+ be enabled on a per-tile basis, instead of on a per tile-column
+ basis. In combination with the new "tile-rows" property, this
+ allows the encoder to make much better use of the available CPU
+ power.
+
+- vp9dec, vp9enc: add support for 10-bit 4:2:0 and 4:2:2 YUV, as well
+ as 8-bit 4:4:4
+
+- vp8enc, vp9enc now default to “good quality” for the deadline
+ property rather then “best quality”. Having the deadline set to best
+ quality causes the encoder to be absurdly slow, most real-life users
+ will prefer good-enough quality with better performance instead.
+
+- wpesrc:
+
+ - implement audio support: a new sometimes source pad will be
+ created for each audio stream created by the web engine.
+ - move wpesrc to wpevideosrc and add a wrapper bin wpesrc to also
+ support audio
+ - also handles web:// URIs now (same as cefsrc)
+ - post messages with the estimated load progress on the bus
-Plugin and library moves
+- x265enc: add negative DTS support, which means timestamps are now
+ offset by 1h same as with x264enc
+
+RTP Payloaders and Depayloaders
+
+- rtpisacpay, rtpisacdepay: new RTP payloader and depayloader for iSAC
+ audio codec
+
+- rtph264depay:
+
+ - new "request-keyframe" property to make the depayloader
+ automatically request a new keyframe from the sender on packet
+ loss, consistent with the new property on rtpvp8depay.
+ - new "wait-for-keyframe" property to make depayloader wait for a
+ new keyframe at the beginning and after packet loss (only
+ effective if the depayloader outputs AUs), consistent with the
+ existing property on rtpvp8depay.
+
+- rtpopuspay, rtpopusdepay: support libwebrtc-compatible multichannel
+ audio in addition to the previously supported multichannel audio
+ modes
+
+- rtpopuspay: add DTX (Discontinuous Transmission) support
+
+- rtpvp8depay: new "request-keyframe" property to make the depayloader
+ automatically request a new keyframe from the sender on packet loss.
+
+- rtpvp8pay: temporal scaling support
-- this section will be filled in in due course
+- rtpvp9depay: Improved SVC handling (aggregate all layers)
+
+RTP Infrastructure
+
+- rtpst2022-1-fecdec, rtpst2022-1-fecenc: new elements providing SMPTE
+ 2022-1 2-D Forward Error Correction. More details in Mathieu’s blog
+ post.
+
+- rtpreddec: BUNDLE support
+
+- rtpredenc, rtpulpfecenc: add support for Transport-wide Congestion
+ Control (TWCC)
+
+- rtpsession: new "twcc-feedback-interval" property to allow RTCP TWCC
+ reports to be scheduled on a timer instead of per marker-bit.
+
+Plugin and library moves
- There were no plugin moves or library moves in this cycle.
The following elements or plugins have been removed:
-- this section will be filled in in due course
-
- The ofa audio fingerprinting plugin has been removed. The MusicIP
database has been defunct for years so this plugin is likely neither
useful nor used by anyone.
Miscellaneous API additions
-- this section will be filled in in due course
+Core
+
+- gst_buffer_new_memdup() is a convenience function for the
+ widely-used gst_buffer_new_wrapped(g_memdup(data,size),size)
+ pattern.
+
+- gst_caps_features_new_single() creates a new single GstCapsFeatures,
+ avoiding the need to use the vararg function with NULL terminator
+ for simple cases.
+
+- gst_element_type_set_skip_documentation() can be used by plugins to
+ signal that certain elements should not be included in the GStreamer
+ plugin documentation. This is useful for plugins where elements are
+ registered dynamically based on hardware capabilities and/or where
+ the available plugins and properties vary from system to system.
+ This is used in the d3d11 plugin for example to ensure that only the
+ list of default elements is advertised in the documentation.
+
+- gst_type_find_suggest_empty_simple() is a new convenience function
+ for typefinders for cases where there’s only a media type and no
+ other fields.
+
+- New API to create elements and set properties at construction time,
+ which is not only convenient, but also allows GStreamer elements to
+ have construct-only properties: gst_element_factory_make_full(),
+ gst_element_factory_make_valist(),
+ gst_element_factory_make_with_properties(),
+ gst_element_factory_create_full(),
+ gst_element_factory_create_valist(),
+ gst_element_factory_create_with_properties().
+
+- GstSharedTaskPool: new “shared” task pool subclass with slightly
+ different default behaviour than the existing GstTaskPool which
+ would create unlimited number of threads for new tasks. The shared
+ taskpool creates up to N threads (default: 1) and then distributes
+ pending tasks to those threads round-robin style, and blocks if no
+ thread is available. It is possible to join tasks. This can be used
+ by plugins to implement simple multi-threaded processing and is used
+ for the new multi-threaded video conversion and compositing done in
+ GstVideoAggregator, videoconverter and compositor.
+
+Plugins Base Utils library
+
+- GstDiscoverer:
+
+ - gst_discoverer_container_info_get_tags() was added to retrieve
+ global/container tags (vs. per-stream tags). Per-Stream tags can
+ be retrieved via the existing
+ gst_discoverer_stream_info_get_tags().
+ gst_discoverer_info_get_tags(), which for many files returns a
+ confusing mix of stream and container tags, has been deprecated
+ in favour of the container/stream-specific functions.
+ - gst_discoverer_stream_info_get_stream_number() returns a unique
+ integer identifier for a given stream within the given
+ GstDiscoverer context. (If this matches the stream number inside
+ the container bitstream that’s by coincidence and not by
+ design.)
+
+- gst_pb_utils_get_caps_description_flags() can be used to query
+ whether certain caps represent a container, audio, video, image,
+ subtitles, tags, or something else. This only works for formats
+ known to GStreamer.
+
+- gst_pb_utils_get_file_extension_from_caps() returns a possible file
+ extension for given caps.
+
+- gst_codec_utils_h264_get_profile_flags_level(): Parses profile,
+ flags, and level from H264 AvcC codec_data. The format of H264 AVCC
+ extradata/sequence_header is documented in the ITU-T H.264
+ specification section 7.3.2.1.1 as well as in ISO/IEC 14496-15
+ section 5.3.3.1.2.
+
+- gst_codec_utils_caps_get_mime_codec() to convert caps to a RFC 6381
+ compatible MIME codec string codec. Useful for providing the codecs
+ field inside the Content-Type HTTP header for containerized formats,
+ such as mp4 or matroska.
+
+GStreamer OpenGL integration library and plugins
+
+- glcolorconvert: added suppport for converting the video formats
+ A420, AV12, BGR, BGRA, RGBP and BGRP.
+
+- Added support to GstGLBuffer for persistent buffer mappings where a
+ Pixel Buffer Object (PBO) can be mapped by both the CPU and the GPU.
+ This removes a memcpy() when uploading textures or vertices
+ particularly when software decoders (e.g. libav) are direct
+ rendering into our memory. Improves transfer performance
+ significantly. Requires OpenGL 4.4, GL_ARB_buffer_storage or
+ GL_EXT_buffer_storage
+
+- Added various helper functions for handling 4x4 matrices of affine
+ transformations as used by GstVideoAffineTransformationMeta.
+
+- Add support to GstGLContext for allowing the application to control
+ the config (EGLConfig, GLXConfig, etc) used when creating the OpenGL
+ context. This allows the ability to choose between RGB16 or RGB10A2
+ or RGBA8 back/front buffer configurations that were previously
+ hardcoded. GstGLContext also supports retrieving the configuration
+ it was created with or from an externally provide OpenGL context
+ handle. This infrastructure is also used to create a compatible
+ config from an application/externally provided OpenGL context in
+ order to improve compatibility with other OpenGL frameworks and GUI
+ toolkits. A new environment variable GST_GL_CONFIG was also added to
+ be able to request a specific configuration from the command line.
+ Note: different platforms will have different functionality
+ available.
+
+- Add support for choosing between EGL and WGL at runtime when running
+ on Windows. Previously this was a build-time switch. Allows use in
+ e.g. Gtk applications on Windows that target EGL/ANGLE without
+ recompiling GStreamer. gst_gl_display_new_with_type() can be used by
+ applications to choose a specific display type to use.
+
+- Build fixes to explicitly check for Broadcom-specific libraries on
+ older versions of the Raspberry Pi platform. The Broadcom OpenGL ES
+ and EGL libraries have different filenames. Using the vc4 Mesa
+ driver on the Raspberry Pi is not affected.
+
+- Added support to glupload and gldownload for transferring RGBA
+ buffers using the memory:NVMM available on the Nvidia Tegra family
+ of embedded devices.
+
+- Added support for choosing libOpenGL and libGLX as used in a GLVND
+ environment on unix-based platforms. This allows using desktop
+ OpenGL and EGL without pulling in any GLX symbols as would be
+ required with libGL.
+
+Video library
+
+- New raw video formats:
+
+ - AV12 (NV12 with alpha plane)
+ - RGBP and BGRP (planar RGB formats)
+ - ARGB64 variants with specified endianness instead of host
+ endianness:
+ - ARGB64_LE, ARGB64_BE
+ - RGBA64_BE, RGBA64_LE
+ - BGRA64_BE, BGRA64_LE
+ - ABGR64_BE, ABGR64_LE
+
+- gst_video_orientation_from_tag() is new convenience API to parse the
+ image orientation from a GstTagList.
+
+- GstVideoDecoder subframe support (see below)
+
+- GstVideoCodecState now also carries some HDR metadata
+
+- Ancillary video data: implement transform functions for AFD/Bar
+ metas, so they will be forwarded in more cases
+
+MPEG-TS library
+
+This library only handles section parsing and such, see above for
+changes to the actual mpegtsmux and mpegtsdemux elements.
+
+- many additions and improvements to SCTE-35 section parsing
+- new API for fetching extended descriptors:
+ gst_mpegts_find_descriptor_with_extension()
+- add support for SIT sections (Selection Information Tables)
+- expose event-from-section constructor gst_event_new_mpegts_section()
+- parse Audio Preselection Descriptor needed for Dolby AC-4
+
+GstWebRTC library + webrtcbin
+
+- Change the way in which sink pads and transceivers are matched
+ together to support easier usage. If a pad is created without a
+ specific index (i.e. using sink_%u as the pad template), then an
+ available compatible transceiver will be searched for. If a specific
+ index is requested (i.e. sink_1) then if a transceiver for that
+ m-line already exists, that transceiver must match the new sink pad
+ request. If there is no transceiver available in either scenario, a
+ new transceiver is created. If a mixture of both sink_1 and sink_%u
+ requests result in an impossible situation, an error will be
+ produced at pad request time or from create offer/answer.
+
+- webrtcbin now uses regular ICE nomination instead of libnice’s
+ default of aggressive ICE nomination. Regular ICE nomination is the
+ default recommended by various relevant standards and improves
+ connectivity in specific network scenarios.
+
+- Add support for limiting the port range used for RTP with the
+ addition of the min-rtp-port and max-rtp-port properties on the ICE
+ object.
+
+- Expose the SCTP transport as a property on webrtcbin to more closely
+ match the WebRTC specification.
+
+- Added support for taking into account the data channel transport
+ state when determining the value of the "connection-state" property.
+ Previous versions of the WebRTC spec did not include the data
+ channel state when computing this value.
+
+- Add configuration for choosing the size of the underlying sockets
+ used for transporting media data
+
+- Always advertise support for the transport-cc RTCP feedback protocol
+ as rtpbin supports it. For full support, the configured caps (input
+ or through codec-preferences) need to include the relevant RTP
+ header extension.
+
+- Numerous fixes to caps and media handling to fail-fast when an
+ incompatible situation is detected.
+
+- Improved support for attaching the required media after a remote
+ offer has been set.
+
+- Add support for dynamically changing the amount of FEC used for a
+ particular stream.
+
+- webrtcbin now stops further SDP processing at the first error it
+ encounters.
+
+- Completed support for either local or the remote closing a data
+ channel.
+
+- Various fixes when performing BUNDLEing of the media streams in
+ relation to RTX and FEC usage.
+
+- Add support for writing out QoS DSCP marking on outgoing packets to
+ improve reliability in some network scenarios.
+
+- Improvements to the statistics returned by the get-stats signal
+ including the addition of the raw statistics from the internal
+ RTPSource, the TWCC stats when available.
+
+- The webrtc library does not expose any objects anymore with public
+ fields. Instead properties have been added to replace that
+ functionality. If you are accessing such fields in your application,
+ switch to the corresponding properties.
+
+GstCodecs and Video Parsers
+
+- Support for render delays to improve throughput across all CODECs
+ (used with NVDEC and V4L2).
+- lots of improvements to parsers and the codec parsing decoder base
+ classes (H264, H265, VP8, VP9, AV1, MPEG-2) used for various
+ hardware-accelerate decoder APIs.
+
+Bindings support
+
+- gst_allocation_params_new() allocates a GstAllocationParams struct
+ on the heap. This should only be used by bindings (and freed via
+ gst_allocation_params_free() then). In C code you would allocate
+ this on the stack and only init it in place.
+
+- gst_debug_log_literal() can be used to log a string to the debug log
+ without going through any printf format expansion and associated
+ overhead. This is mostly useful for bindings such as the Rust
+ bindings which may have done their own formatting already .
+
+- Provide non-inlined versions of refcounting APIs for various
+ GStreamer mini objects, so that they can be consumed by bindings
+ (e.g. gstreamer-sharp): gst_buffer_ref, gst_buffer_unref,
+ gst_clear_buffer, gst_buffer_copy, gst_buffer_replace,
+ gst_buffer_list_ref, gst_buffer_list_unref, gst_clear_buffer_list,
+ gst_buffer_list_copy, gst_buffer_list_replace, gst_buffer_list_take,
+ gst_caps_ref, gst_caps_unref, gst_clear_caps, gst_caps_replace,
+ gst_caps_take, gst_context_ref, gst_context_unref, gst_context_copy,
+ gst_context_replace, gst_event_replace, gst_event_steal,
+ gst_event_take, gst_event_ref, gst_event_unref, gst_clear_event,
+ gst_event_copy, gst_memory_ref, gst_memory_unref, gst_message_ref,
+ gst_message_unref, gst_clear_message, gst_message_copy,
+ gst_message_replace, gst_message_take, gst_promise_ref,
+ gst_promise_unref, gst_query_ref, gst_query_unref, gst_clear_query,
+ gst_query_copy, gst_query_replace, gst_query_take, gst_sample_ref,
+ gst_sample_unref, gst_sample_copy, gst_tag_list_ref,
+ gst_tag_list_unref, gst_clear_tag_list, gst_tag_list_replace,
+ gst_tag_list_take, gst_uri_copy, gst_uri_ref, gst_uri_unref,
+ gst_clear_uri.
+
+- expose a GType for GstMiniObject
+
+- gst_device_provider_probe() now returns non-floating device object
+
+API Deprecations
+
+- gst_element_get_request_pad() has been deprecated in favour of the
+ newly-added gst_element_request_pad_simple() which does the exact
+ same thing but has a less confusing name that hopefully makes clear
+ that the function request a new pad rather than just retrieves an
+ already-existing request pad.
+
+- gst_discoverer_info_get_tags(), which for many files returns a
+ confusing mix of stream and container tags, has been deprecated in
+ favour of the container-specific and stream-specific functions,
+ gst_discoverer_container_info_get_tags() and
+ gst_discoverer_stream_info_get_tags().
+
+- gst_video_sink_center_rect() was deprecated in favour of the more
+ generic newly-added gst_video_center_rect().
+
+- The GST_MEMORY_FLAG_NO_SHARE flag has been deprecated, as it tends
+ to cause problems and prevents sub-buffering. If pooling or lifetime
+ tracking is required, memories should be allocated through a custom
+ GstAllocator instead of relying on the lifetime of the buffers the
+ memories were originally attached to, which is fragile anyway.
+
+- The GstPlayer high-level playback library is being replaced with the
+ new GstPlay library (see above). GstPlayer should be considered
+ deprecated at this point and will be marked as such in the next
+ development cycle. Applications should be ported to GstPlay.
+
+- Gstreamer Editing Services: ges_video_transition_set_border(),
+ ges_video_transition_get_border()
+ ges_video_transition_set_inverted()
+ ges_video_transition_is_inverted() have been deprecated, use
+ ges_timeline_element_set_children_properties() instead.
Miscellaneous performance, latency and memory optimisations
-- this section will be filled in in due course
+More video conversion fast paths
-Miscellaneous other changes and enhancements
+- v210 ↔ I420, YV12, Y42B, UYVY and YUY2
+- A420 → RGB
+
+Less jitter when waiting on the system clock
+
+- Better system clock wait accuracy, less jitter: where available,
+ clock_nanosleep is used for higher accuracy for waits below 500
+ usecs, and waits below 2ms will first use the regular waiting system
+ and then clock_nanosleep for the remainder. The various wait
+ implementation have a latency ranging from 50 to 500+ microseconds.
+ While this is not a major issue when dealing with a low number of
+ waits per second (for ex: video), it does introduce a non-negligible
+ jitter for synchronisation of higher packet rate systems.
+
+Video decoder subframe support
-- this section will be filled in in due course
+- The GstVideoDecoder base class gained API to process input at the
+ sub-frame level. That way video decoders can start decoding slices
+ before they have received the full input frame in its entirety (to
+ the extent this is supported by the codec, of course). This helps
+ with CPU utilisation and reduces latency.
-Tracing framework and debugging improvements
+- This functionality is now being used in the OpenJPEG JPEG 2000
+ decoder, the FFmpeg H.264 decoder (in case of NAL-aligned input) and
+ the OpenMAX H.264/H.265 decoders (in case of NAL-aligned input).
-- this section will be filled in in due course
+Miscellaneous other changes and enhancements
+
+- GstDeviceMonitor no longer fails to start just because one of the
+ device providers failed to start. That could happen for example on
+ systems where the pulseaudio device provider is installed, but
+ pulseaudio isn’t actually running but ALSA is used for audio
+ instead. In the same vein the device monitor now keeps track of
+ which providers have been started (via the new
+ gst_device_provider_is_started()) and only stops actually running
+ device providers when stopping the device monitor.
+
+- On embedded systems it can be useful to create a registry that can
+ be shared and read by multiple processes running as different users.
+ It is now possible to set the new GST_REGISTRY_MODE environment
+ variable to specify the file mode for the registry file, which by
+ default is set to be only user readable/writable.
+
+- GstNetClientClock will signal lost sync in case the remote time
+ resets (e.g. because device power cycles), by emitting the “synced”
+ signal with synced=FALSE parameter, so applications can take action.
+
+- gst_value_deserialize_with_pspec() allows deserialization with a
+ hint for what the target GType should be. This allows for example
+ passing arrays of flags through the command line or
+ gst_util_set_object_arg(), eg: foo="<bar,bar+baz>".
+
+- It’s now allowed to create an empty GstVideoOverlayComposition
+ without any rectangles by passing a NULL rectangle to
+ gst_video_overlay_composition_new(). This is useful for bindings and
+ simplifies application code in some places.
+
+Tracing framework, debugging and testing improvements
+
+- New factories tracer to list loaded elements (and other plugin
+ features). This can be useful to collect a list of elements needed
+ for an application, which then in turn can be used to create a
+ tailored minimal GStreamer build that contains just the elements
+ needed and nothing else.
+- New plugin-feature-loaded tracing hook for use by tracers like the
+ new factories tracer
+
+- GstHarness: Add gst_harness_set_live() so that harnesses can be set
+ to non-live and return is-live=false in latency queries if needed.
+ Default behaviour is to always return is-live=true in latency
+ queries.
+
+- navseek: new "hold-eos" property. When enabled, the element will
+ hold back an EOS event until the next keystroke (via navigation
+ events). This can be used to keep a video sink showing the last
+ frame of a video pipeline until a key is pressed instead of tearing
+ it down immediately on EOS.
+
+- New fakeaudiosink element: mimics an audio sink and can be used for
+ testing and CI pipelines on systems where no audio system is
+ installed or running. It differs from fakesink in that it only
+ support audio caps and syncs to the clock by default like a normal
+ audio sink. It also implements the GstStreamVolume interface like
+ most audio sinks do.
+
+- New videocodectestsink element for video codec conformance testing:
+ Calculates MD5 checksums for video frames and skips any padding
+ whilst doing so. Can optionally also write back the video data with
+ padding removed into a file for easy byte-by-byte comparison with
+ reference data.
Tools
-- this section will be filled in in due course
+gst-inspect-1.0
+
+- Can sort the list of plugins by passing --sort=name as command line
+ option
+
+gst-launch-1.0
+
+- will now error out on top-level properties that don’t exist and
+ which were silently ignored before
+- On Windows the high-resolution clock is enabled now, which provides
+ better clock and timer performance on Windows (see Windows section
+ below for more details).
+
+gst-play-1.0
+
+- New --start-position command line argument to start playback from
+ the specified position
+- Audio can be muted/unmuted in interactive mode by pressing the m
+ key.
+- On Windows the high-resolution clock is enabled now (see Windows
+ section below for more details)
+
+gst-device-monitor-1.0
+
+- New --include-hidden command line argument to also show “hidden”
+ device providers
+
+ges-launch-1.0
+
+- New interactive mode that allows seeking and such. Can be disabled
+ by passing the --no-interactive argument on the command line.
+- Option to forward tags
+- Allow using an existing clip to determine the rendering format (both
+ topology and profile) via new --profile-from command line argument.
GStreamer RTSP server
-- this section will be filled in in due course
+- GstRTSPMediaFactory gained API to disable RTCP
+ (gst_rtsp_media_factory_set_enable_rtcp(), "enable-rtcp" property).
+ Previously RTCP was always allowed for all RTSP medias. With this
+ change it is possible to disable RTCP completely, no matter if the
+ client wants to do RTCP or not.
+
+- Make a mount point of / work correctly. While not allowed by the
+ RTSP 2 spec, the RTSP 1 spec is silent on this and it is used in the
+ wild. It is now possible to use / as a mount path in
+ gst-rtsp-server, e.g. rtsp://example.com/ would work with this now.
+ Note that query/fragment parts of the URI are not necessarily
+ correctly handled, and behaviour will differ between various
+ client/server implementations; so use it if you must but don’t bug
+ us if it doesn’t work with third party clients as you’d hoped.
+
+- multithreading fixes (races, refcounting issues, deadlocks)
+
+- ONVIF audio backchannel fixes
+
+- ONVIF trick mode optimisations
+
+- rtspclientsink: new "update-sdp" signal that allows updating the SDP
+ before sending it to the server via ANNOUNCE. This can be used to
+ add additional metadata to the SDP, for example. The order and
+ number of medias must not be changed, however.
GStreamer VAAPI
-- this section will be filled in in due course
+- new AV1 decoder element (vaapiav1dec)
+
+- H264 decoder: handle stereoscopic 3D video with frame packing
+ arrangement SEI messages
+
+- H265 encoder: added Screen Content Coding extensions support
+
+- H265 decoder: gained MAIN_444_12 profile support (decoded to
+ Y412_LE), and 4:2:2 12-bits support (decoded to Y212_LE)
+
+- vaapipostproc: gained BT2020 color standard support
+
+- vaapidecode: now generates caps templates dynamically at runtime in
+ order to advertise actually supported caps instead of all
+ theoretically supported caps.
+
+- GST_VAAPI_DRM_DEVICE environment variable to force a specified DRM
+ device when a DRM display is used. It is ignored when other types of
+ displays are used. By default /dev/dri/renderD128 is used for DRM
+ display.
GStreamer OMX
-- this section will be filled in in due course
+- subframe support in H.264/H.265 decoders
GStreamer Editing Services and NLE
-- this section will be filled in in due course
+- framepositioner: new "operator" property to access blending modes in
+ the compositor
+- timeline: Implement snapping to markers
+- smart-mixer: Add support for d3d11compositor and glvideomixer
+- titleclip: add "draw-shadow" child property
+- ges:// URI support to define a timeline from a description.
+- command-line-formatter
+ - Add track management to timeline description
+ - Add keyframe support
+- ges-launch-1.0:
+ - Add an interactive mode where we can seek etc…
+ - Add option to forward tags
+ - Allow using an existing clip to determine the rendering format
+ (both topology and profile) via new --profile-from command line
+ argument.
+- Fix static build
GStreamer validate
-- this section will be filled in in due course
+- report: Add a way to force backtraces on reports even if not a
+ critical issue (GST_VALIDATE_ISSUE_FLAGS_FORCE_BACKTRACE)
+- Add a flag to gst_validate_replace_variables_in_string() allow
+ defining how to resolve variables in structs
+- Add gst_validate_bin_monitor_get_scenario() to get the bin monitor
+ scenario, which is useful for applications that use Validate
+ directly.
+- Add an expected-values parameter to wait, message-type=XX allowing
+ more precise filtering of the message we are waiting for.
+- Add config file support: each test can now use a config file for the
+ given media file used to test.
+- Add support to check properties of object properties
+- scenario: Add an "action-done" signal to signal when an action is
+ done
+- scenario: Add a "run-command" action type
+- scenario: Allow forcing running action on idle from scenario file
+- scenario: Allow iterating over arrays in foreach
+- scenario: Rename ‘interlaced’ action to ‘non-blocking’
+- scenario: Add a non-blocking flag to the wait signal
GStreamer Python Bindings
-- this section will be filled in in due course
+- Fixes for Python 3.10
+- Various build fixes
+- at least one known breaking change caused by g-i annotation changes
+ (see below)
GStreamer C# Bindings
-- this section will be filled in in due course
+- Fix GstDebugGraphDetails enum
+- Updated to latests GtkSharp
+- Updated to include GStreamer 1.20 API
GStreamer Rust Bindings and Rust Plugins
-The GStreamer Rust bindings are released separately with a different
-release cadence that’s tied to gtk-rs, but the latest release has
-already been updated for the upcoming new GStreamer 1.20 API.
-
-gst-plugins-rs, the module containing GStreamer plugins written in Rust,
-has also seen lots of activity with many new elements and plugins.
-
-What follows is a list of elements and plugins available in
-gst-plugins-rs, so people don’t miss out on all those potentially useful
-elements that have no C equivalent.
-
-- FIXME: add new elements
-
-Rust audio plugins
-
-- audiornnoise: New element for audio denoising which implements the
- noise removal algorithm of the Xiph RNNoise library, in Rust
-- rsaudioecho: Port of the audioecho element from gst-plugins-good
- rsaudioloudnorm: Live audio loudness normalization element based on
- the FFmpeg af_loudnorm filter
-- claxondec: FLAC lossless audio codec decoder element based on the
- pure-Rust claxon implementation
-- csoundfilter: Audio filter that can use any filter defined via the
- Csound audio programming language
-- lewtondec: Vorbis audio decoder element based on the pure-Rust
- lewton implementation
-
-Rust video plugins
-
-- cdgdec/cdgparse: Decoder and parser for the CD+G video codec based
- on a pure-Rust CD+G implementation, used for example by karaoke CDs
-- cea608overlay: CEA-608 Closed Captions overlay element
-- cea608tott: CEA-608 Closed Captions to timed-text (e.g. VTT or SRT
- subtitles) converter
-- tttocea608: CEA-608 Closed Captions from timed-text converter
-- mccenc/mccparse: MacCaption Closed Caption format encoder and parser
-- sccenc/sccparse: Scenarist Closed Caption format encoder and parser
-- dav1dec: AV1 video decoder based on the dav1d decoder implementation
- by the VLC project
-- rav1enc: AV1 video encoder based on the fast and pure-Rust rav1e
- encoder implementation
-- rsflvdemux: Alternative to the flvdemux FLV demuxer element from
- gst-plugins-good, not feature-equivalent yet
-- rsgifenc/rspngenc: GIF/PNG encoder elements based on the pure-Rust
- implementations by the image-rs project
-
-Rust text plugins
-
-- textwrap: Element for line-wrapping timed text (e.g. subtitles) for
- better screen-fitting, including hyphenation support for some
- languages
-
-Rust network plugins
-
-- reqwesthttpsrc: HTTP(S) source element based on the Rust
- reqwest/hyper HTTP implementations and almost feature-equivalent
- with the main GStreamer HTTP source souphttpsrc
-- s3src/s3sink: Source/sink element for the Amazon S3 cloud storage
-- awstranscriber: Live audio to timed text transcription element using
- the Amazon AWS Transcribe API
-
-Generic Rust plugins
-
-- sodiumencrypter/sodiumdecrypter: Encryption/decryption element based
- on libsodium/NaCl
-- togglerecord: Recording element that allows to pause/resume
- recordings easily and considers keyframe boundaries
-- fallbackswitch/fallbacksrc: Elements for handling potentially
- failing (network) sources, restarting them on errors/timeout and
- showing a fallback stream instead
-- threadshare: Set of elements that provide alternatives for various
- existing GStreamer elements but allow to share the streaming threads
- between each other to reduce the number of threads
-- rsfilesrc/rsfilesink: File source/sink elements as replacements for
- the existing filesrc/filesink elements
+- The GStreamer Rust bindings are released separately with a different
+ release cadence that’s tied to gtk-rs, but the latest release has
+ already been updated for the upcoming new GStreamer 1.20 API (v1_20
+ feature).
+
+- gst-plugins-rs, the module containing GStreamer plugins written in
+ Rust, has also seen lots of activity with many new elements and
+ plugins. See the New Elements section above for a list of new Rust
+ elements.
Build and Dependencies
- Meson 0.59 or newer is required to build GStreamer now.
-- FIXME: this section will be filled in in due course
+- The GLib requirement has been bumped to GLib 2.56 or newer (from
+ March 2018).
+
+- The wpe plugin now requires wpe >= 2.28 and wpebackend-fdo >= 1.8
Explicit opt-in required for build of certain plugins with (A)GPL dependencies
See Building plugins with (A)GPL-licensed dependencies for more details
and a non-exhaustive list of plugins affected.
-gst-build: replaced by Monorepo
+gst-build: replaced by mono repository
-- this section will be filled in in due course
-
-- FIXME: describe + link to Monorepo FAQ
+See mono repository section above and the GStreamer mono repository FAQ.
Cerbero
on platforms where dependencies are not readily available, such as
Windows, Android, iOS and macOS.
-General improvements
+General Cerbero improvements
-- this section will be filled in in due course
+- Plugin removed: libvisual
+- New plugins: rtpmanagerbad and rist
-macOS / iOS
+macOS / iOS specific Cerbero improvements
-- this section will be filled in in due course
+- XCode 12 support
+- macOS OS release support is now future-proof, similar to iOS
+- macOS Apple Silicon (ARM64) cross-compile support has been added
+- macOS Apple Silicon (ARM64) native support is currently experimental
-Windows
+Windows specific Cerbero improvements
-- this section will be filled in in due course
+- Visual Studio 2022 support has been added
+- bootstrap is faster since it requires building fewer build-tools
+ recipes on Windows
+- package is faster due to better scheduling of recipe stages and
+ elimination of unnecessary autotools regeneration
+- The following plugins are no longer built on Windows:
+ - a52dec (another decoder is still available in libav)
+ - dvdread
+ - resindvd
Windows MSI installer
-- this section will be filled in in due course
+- no major changes
-Linux
+Linux specific Cerbero improvements
-- this section will be filled in in due course
+- Fedora, Debian OS release support is now more future-proof
+- Amazon Linux 2 support has been added
-Android
+Android specific Cerbero improvements
-- this section will be filled in in due course
+- no major changes
Platform-specific changes and improvements
Android
-- this section will be filled in in due course
+- No major changes
macOS and iOS
-- this section will be filled in in due course
+- applemedia: add ProRes support to vtenc and vtdec
Windows
-- this section will be filled in in due course
+- On Windows the high-resolution clock is enabled now in the
+ gst-launch-1.0 and gst-play-1.0 command line tools, which provides
+ better clock and timer performance on Windows, at the cost of higher
+ power consumption. By default, without the high-resolution clock
+ enabled, the timer precision on Windows is system-dependent and may
+ be as bad as 15ms which is not good enough for many multimedia
+ applications. Developers may want to do the same in their Windows
+ applications if they think it’s a good idea for their application
+ use case, and depending on the Windows version they target. This is
+ not done automatically by GStreamer because on older Windows
+ versions (pre-Windows 10) this affects a global Windows setting and
+ also there’s a power consumption vs. performance trade-off that may
+ differ from application to application.
+
+- dxgiscreencapsrc now supports resolution changes
+
+- The wasapi2 audio plugin was rewritten and now has a higher rank
+ than the old wasapi plugin since it has a number of additional
+ features such as automatic stream routing, and no
+ known-but-hard-to-fix issues. The plugin is always built if the
+ Windows 10 SDK is available now.
+
+- The wasapi device providers now detect and notify dynamic device
+ additions/removals
+
+- d3d11screencapturesrc: new desktop capture element, including
+ GstDeviceProvider implementation to enumerate/select target monitors
+ for capture.
+
+- Direct3D11/DXVA decoder now supports AV1 and MPEG2 codecs
+ (d3d11av1dec, d3d11mpeg2dec)
+
+- VP9 decoding got more reliable and stable thanks to a newly written
+ codec parser
+
+- Support for decoding interlaced H.264/AVC streams
+
+- Hardware-accelerated video deinterlacing (d3d11deinterlace) and
+ video mixing (d3d11compositor)
+
+- Video mixing with the Direct3D11 API (d3d11compositor)
+
+- MediaFoundation API based hardware encoders gained the ability to
+ receive Direct3D11 textures as an input
+
+- Seungha’s blog post “GStreamer ❤ Windows: A primer on the cool stuff
+ you’ll find in the 1.20 release” describes many of the
+ Windows-related improvements in more detail
Linux
-- this section will be filled in in due course
+- bluez: LDAC Bluetooth audio codec support in a2dpsink and avdtpsink,
+ as well as an LDAC RTP payloader (rtpldacpay) and an LDAC audio
+ encoder (ldacenc)
+
+- kmssink: gained support for NV24, NV61, RGB16/BGR16 formats;
+ auto-detect NVIDIA Tegra driver
Documentation improvements
-- this section will be filled in in due course
+- hardware-accelerated GPU plugins will now no longer always list all
+ the element variants for all available GPUs, since those are
+ system-dependent and it’s confusing for users to see those in the
+ documentation just because the GStreamer developer who generated the
+ docs had multiple GPUs to play with at the time. Instead just show
+ the default elements.
-Possibly Breaking Changes
+Possibly Breaking and Other Noteworthy Behavioural Changes
+
+- gst_parse_launch(), gst_parse_bin_from_description() and friends
+ will now error out when setting properties that don’t exist on
+ top-level bins. They were silently ignored before.
+
+- The GstWebRTC library does not expose any objects anymore with
+ public fields. Instead properties have been added to replace that
+ functionality. If you are accessing such fields in your application,
+ switch to the corresponding properties.
-- FIXME: this section will be filled in in due course
-- MPEG-TS SCTE-35 API changes (FIXME: flesh out)
-- gst_parse_launch() and friends now error out on non-existing
- properties on top-level bins where they would silently fail and
- ignore those before.
- playbin and uridecodebin now emit the source-setup signal before the
element is added to the bin and linked so that the source element is
already configured before any scheduling query comes in, which is
- useful for elements such as appsrc or giostreamsrc. (Merge Request)
+ useful for elements such as appsrc or giostreamsrc.
+
+- The source element inside urisourcebin (used inside uridecodebin3
+ which is used inside playbin3) is no longer called "source". This
+ shouldn’t affect anyone hopefully, because there’s a "setup-source"
+ signal to configure the source element and no one should rely on
+ names of internal elements anyway.
+
+- The vp8enc element now expects bps (bits per second) for the
+ "temporal-scalability-target-bitrate" property, which is consistent
+ with the "target-bitrate" property. Since additional configuration
+ is required with modern libvpx to make temporal scaling work anyway,
+ chances are that very few people will have been using this property
+
+- vp8enc and vp9enc now default to “good quality” for the "deadline"
+ property rather then “best quality”. Having the deadline set to best
+ quality causes the encoder to be absurdly slow, most real-life users
+ will want the good quality tradeoff instead.
+
+- The experimental GstTranscoder library API in gst-plugins-bad was
+ changed from a GObject signal-based notification mechanism to a
+ GstBus/message-based mechanism akin to GstPlayer/GstPlay.
+
+- MPEG-TS SCTE-35 API: semantic change for SCTE-35 splice commands:
+ timestamps passed by the application should be in running time now,
+ since users of the API can’t really be expected to predict the local
+ PTS of the muxer.
+
+- The GstContext used by souphttpsrc to share the session between
+ multiple element instances has changed. Previously it provided
+ direct access to the internal SoupSession object, now it only
+ provides access to an opaque, internal type. This change is
+ necessary because SoupSession is not thread-safe at all and can’t be
+ shared safely between arbitrary external code and souphttpsrc.
+
+- Python bindings: GObject-introspection related Annotation fixes have
+ led to a case of a GstVideo.VideoInfo-related function signature
+ changing in the Python bindings (possibly one or two other cases
+ too). This is for a function that should never have been exposed in
+ the first place though, so the bindings are being updated to throw
+ an exception in that case, and the correct replacement API has been
+ added in form of an override.
Known Issues
-- this section will be filled in in due course
-
-- There are a couple of known WebRTC-related regressions/blockers:
-
- - webrtc: DTLS setup with Chrome is broken
- - webrtcbin: First keyframe is usually lost
+- nothing in particular at this point (but also see possibly breaking
+ changes section above)
Contributors
-- this section will be filled in in due course
+Aaron Boxer, Adam Leppky, Adam Williamson, Alba Mendez, Alejandro
+González, Aleksandr Slobodeniuk, Alexander Vandenbulcke, Alex Ashley,
+Alicia Boya García, Andika Triwidada, Andoni Morales Alastruey, Andrew
+Wesie, Andrey Moiseev, Antonio Ospite, Antonio Rojas, Arthur Crippa
+Búrigo, Arun Raghavan, Ashley Brighthope, Axel Kellermann, Baek, Bastien
+Nocera, Bastien Reboulet, Benjamin Gaignard, Bing Song, Binh Truong,
+Biswapriyo Nath, Brad Hards, Brad Smith, Brady J. Garvin, Branko
+Subasic, Camilo Celis Guzman, Chris Bass, ChrisDuncanAnyvision, Chris
+White, Corentin Damman, Daniel Almeida, Daniel Knobe, Daniel Stone,
+david, David Fernandez, David Keijser, David Phung, Devarsh Thakkar,
+Dinesh Manajipet, Dmitry Samoylov, Dmitry Shusharin, Dominique Martinet,
+Doug Nazar, Ederson de Souza, Edward Hervey, Emmanuel Gil Peyrot,
+Enrique Ocaña González, Ezequiel Garcia, Fabian Orccon, Fabrice
+Fontaine, Fernando Jimenez Moreno, Florian Karydes, Francisco Javier
+Velázquez-García, François Laignel, Frederich Munch, Fredrik Pålsson,
+George Kiagiadakis, Georg Lippitsch, Göran Jönsson, Guido Günther,
+Guillaume Desmottes, Guiqin Zou, Haakon Sporsheim, Haelwenn (lanodan)
+Monnier, Haihao Xiang, Haihua Hu, Havard Graff, He Junyan, Helmut
+Januschka, Henry Wilkes, Hosang Lee, Hou Qi, Ignacio Casal Quinteiro,
+Igor Kovalenko, Ilya Kreymer, Imanol Fernandez, Jacek Tomaszewski, Jade
+Macho, Jakub Adam, Jakub Janků, Jan Alexander Steffens (heftig), Jan
+Schmidt, Jason Carrete, Jason Pereira, Jay Douglass, Jeongki Kim, Jérôme
+Laheurte, Jimmi Holst Christensen, Johan Sternerup, John Hassell, John
+Lindgren, John-Mark Bell, Jonathan Matthew, Jordan Petridis, Jose
+Quaresma, Julian Bouzas, Julien, Kai Uwe Broulik, Kasper Steensig
+Jensen, Kellermann Axel, Kevin Song, Khem Raj, Knut Inge Hvidsten, Knut
+Saastad, Kristofer Björkström, Lars Lundqvist, Lawrence Troup, Lim Siew
+Hoon, Lucas Stach, Ludvig Rappe, Luis Paulo Fernandes de Barros, Luke
+Yelavich, Mads Buvik Sandvei, Marc Leeman, Marco Felsch, Marek Vasut,
+Marian Cichy, Marijn Suijten, Marius Vlad, Markus Ebner, Mart Raudsepp,
+Matej Knopp, Mathieu Duponchelle, Matthew Waters, Matthieu De Beule,
+Mengkejiergeli Ba, Michael de Gans, Michael Olbrich, Michael Tretter,
+Michal Dzik, Miguel Paris, Mikhail Fludkov, mkba, Nazar Mokrynskyi,
+Nicholas Jackson, Nicola Murino, Nicolas Dufresne, Niklas Hambüchen,
+Nikolay Sivov, Nirbheek Chauhan, Olivier Blin, Olivier Crete, Olivier
+Crête, Paul Goulpié, Per Förlin, Peter Boba, P H, Philippe Normand,
+Philipp Zabel, Pieter Willem Jordaan, Piotrek Brzeziński, Rafał
+Dzięgiel, Rafostar, raghavendra, Raghavendra, Raju Babannavar, Raleigh
+Littles III, Randy Li, Randy Li (ayaka), Ratchanan Srirattanamet, Raul
+Tambre, reed.lawrence, Ricky Tang, Robert Rosengren, Robert Swain, Robin
+Burchell, Roman Sivriver, R S Nikhil Krishna, Ruben Gonzalez, Ruslan
+Khamidullin, Sanchayan Maity, Scott Moreau, Sebastian Dröge, Sergei
+Kovalev, Seungha Yang, Sid Sethupathi, sohwan.park, Sonny Piers, Staz M,
+Stefan Brüns, Stéphane Cerveau, Stephan Hesse, Stian Selnes, Stirling
+Westrup, Théo MAILLART, Thibault Saunier, Tim, Timo Wischer, Tim-Philipp
+Müller, Tim Schneider, Tobias Ronge, Tom Schoonjans, Tulio Beloqui,
+tyler-aicradle, U. Artie Eoff, Ung, Val Doroshchuk, VaL Doroshchuk,
+Víctor Manuel Jáquez Leal, Vivek R, Vivia Nikolaidou, Vivienne
+Watermeier, Vladimir Menshakov, Will Miller, Wim Taymans, Xabier
+Rodriguez Calvar, Xavier Claessens, Xℹ Ruoyao, Yacine Bandou, Yinhang
+Liu, youngh.lee, youngsoo.lee, yychao, Zebediah Figura, Zhang yuankun,
+Zhang Yuankun, Zhao, Zhao Zhili, , Aleksandar Topic, Antonio Ospite,
+Bastien Nocera, Benjamin Gaignard, Brad Hards, Carlos Falgueras García,
+Célestin Marot, Corentin Damman, Corentin Noël, Daniel Almeida, Daniel
+Knobe, Danny Smith, Dave Piché, Dmitry Osipenko, Fabrice Fontaine,
+fjmax, Florian Zwoch, Guillaume Desmottes, Haihua Hu, Heinrich Kruger,
+He Junyan, Jakub Adam, James Cowgill, Jan Alexander Steffens (heftig),
+Jean Felder, Jeongki Kim, Jiri Uncovsky, Joe Todd, Jordan Petridis,
+Krystian Wojtas, Marc-André Lureau, Marcin Kolny, Marc Leeman, Mark
+Nauwelaerts, Martin Reboredo, Mathieu Duponchelle, Matthew Waters,
+Mengkejiergeli Ba, Michael Gruner, Nicolas Dufresne, Nirbheek Chauhan,
+Olivier Crête, Philippe Normand, Rafał Dzięgiel, Ralf Sippl, Robert
+Mader, Sanchayan Maity, Sangchul Lee, Sebastian Dröge, Seungha Yang,
+Stéphane Cerveau, Teh Yule Kim, Thibault Saunier, Thomas Klausner, Timo
+Wischer, Tim-Philipp Müller, Tobias Reineke, Tomasz Andrzejak, Trung Do,
+Tyler Compton, Ung, Víctor Manuel Jáquez Leal, Vivia Nikolaidou, Wim
+Taymans, wngecn, Wonchul Lee, wuchang li, Xavier Claessens, Xi Ruoyao,
+Yoshiharu Hirose, Zhao,
… and many others who have contributed bug reports, translations, sent
suggestions or helped testing.
1.20.0
-1.20.0 is scheduled to be released around October/November 2021.
+1.20.0 is scheduled to be released around early February 2022.
Schedule for 1.22
unstable development version leading up to the stable 1.22 release. The
development of 1.21/1.22 will happen in the git main branch.
-The plan for the 1.22 development cycle is yet to be confirmed.
+The plan for the 1.22 development cycle is yet to be confirmed. Assuming
+no major project-wide reorganisations in the 1.22 cycle we might try and
+aim for a release around August 2022.
1.22 will be backwards-compatible to the stable 1.20, 1.18, 1.16, 1.14,
1.12, 1.10, 1.8, 1.6, 1.4, 1.2 and 1.0 release series.
------------------------------------------------------------------------
These release notes have been prepared by Tim-Philipp Müller with
-contributions from …
+contributions from Matthew Waters, Nicolas Dufresne, Nirbheek Chauhan,
+Sebastian Dröge and Seungha Yang.
License: CC BY-SA 4.0
-This is GStreamer gst-devtools 1.19.3.
+This is GStreamer gst-devtools 1.19.90.
GStreamer 1.19 is the development branch leading up to the next major
stable version which will be 1.20.
<release>
<Version>
+ <revision>1.19.90</revision>
+ <branch>main</branch>
+ <name></name>
+ <created>2022-01-28</created>
+ <file-release rdf:resource="https://gstreamer.freedesktop.org/src/gst-devtools/gst-devtools-1.19.90.tar.xz" />
+ </Version>
+ </release>
+
+ <release>
+ <Version>
<revision>1.19.3</revision>
<branch>main</branch>
<name></name>
project('gst-devtools', 'c',
- version : '1.19.3.1',
+ version : '1.19.90',
meson_version : '>= 0.59',
default_options : [ 'warning_level=1',
'c_std=gnu99',
project('GStreamer manuals and tutorials', 'c',
- version: '1.19.3.1',
+ version: '1.19.90',
meson_version : '>= 0.59')
hotdoc_p = find_program('hotdoc')
+=== release 1.19.90 ===
+
+2022-01-28 14:28:35 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * NEWS:
+ * RELEASE:
+ * gst-editing-services.doap:
+ * meson.build:
+ Release 1.19.90
+
+2022-01-28 14:28:28 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * ChangeLog:
+ Update ChangeLogs for 1.19.90
+
2022-01-24 19:44:32 +0200 Sebastian Dröge <sebastian@centricular.com>
* ges/meson.build:
GStreamer 1.20 Release Notes
GStreamer 1.20 has not been released yet. It is scheduled for release in
-November 2021.
+late January / early February 2022.
1.19.x is the unstable development version that is being developed in
-the git main branch and which will eventually result in 1.20, and 1.19.3
-is the current development release in that series
-
-It is expected that feature freeze will be in early October 2021,
-followed by one or two 1.19.9x pre-releases and the new 1.20 stable
-release around October/November 2021.
+the git main branch and which will eventually result in 1.20, and
+1.19.90 is the first release candidate in that series (1.20rc1).
1.20 will be backwards-compatible to the stable 1.18, 1.16, 1.14, 1.12,
-1.10, 1.8, 1.6,, 1.4, 1.2 and 1.0 release series.
+1.10, 1.8, 1.6, 1.4, 1.2 and 1.0 release series.
See https://gstreamer.freedesktop.org/releases/1.20/ for the latest
version of this document.
-Last updated: Monday 1 November 2021, 01:00 UTC (log)
+Last updated: Wednesday 26 January 2022, 01:00 UTC (log)
Introduction
Highlights
-- this section will be completed in due course
+- Development in GitLab was switched to a single git repository
+ containing all the modules
+- GstPlay: new high-level playback library, replaces GstPlayer
+- WebM Alpha decoding support
+- Encoding profiles can now be tweaked with additional
+ application-specified element properties
+- Compositor: multi-threaded video conversion and mixing
+- RTP header extensions: unified support in RTP depayloader and
+ payloader base classes
+- SMPTE 2022-1 2-D Forward Error Correction support
+- Smart encoding (passthrough) support for VP8, VP9, H.265 in
+ encodebin and transcodebin
+- Runtime compatibility support for libsoup2 and libsoup3 (libsoup3
+ support experimental)
+- Video decoder subframe support
+- Video decoder automatic packet-loss, data corruption, and keyframe
+ request handling for RTP / WebRTC / RTSP
+- MP4 and Matroska muxers now support profile/level/resolution changes
+ for H264/H265 input streams (i.e. codec data changing on the fly)
+- MP4 muxing mode that initially creates a fragmented mp4 which is
+ converted to a regular mp4 on EOS
+- Audio support for the WebKit Port for Embedded (WPE) web page source
+ element
+- CUDA based video color space convert and rescale elements and
+ upload/download elements
+- NVIDIA memory:NVMM support for OpenGL glupload and gldownload
+ elements
+- Many WebRTC improvements
+- The new VA-API plugin implemention fleshed out with more decoders
+ and new postproc elements
+- AppSink API to retrieve events in addition to buffers and buffer
+ lists
+- AppSrc gained more configuration options for the internal queue
+ (leakiness, limits in buffers and time, getters to read current
+ levels)
+- Updated Rust bindings and many new Rust plugins
+- Improved support for custom minimal GStreamer builds
+- Support build against FFmpeg 5.0
+- Linux Stateless CODEC support gained MPEG2 and VP9
+- Windows Direct3D11/DXVA decoder gained AV1 and MPEG2 support
+- Lots of new plugins, features, performance improvements and bug
+ fixes
Major new features and changes
Noteworthy new features and API
-- this section will be filled in in due course
+- gst_element_get_request_pad() has been deprecated in favour of the
+ newly-added gst_element_request_pad_simple() which does the exact
+ same thing but has a less confusing name that hopefully makes clear
+ that the function request a new pad rather than just retrieves an
+ already-existing request pad.
+
+Development in GitLab was switched to a single git repository containing all the modules
+
+The GStreamer multimedia framework is a set of libraries and plugins
+split into a number of distinct modules which are released independently
+and which have so far been developed in separate git repositories in
+freedesktop.org GitLab.
+
+In addition to these separate git repositories there was a gst-build
+module that would use the Meson build systems’s subproject feature to
+download each individual module and then build everything in one go. It
+would also provide an uninstalled development environment that made it
+easy to work on GStreamer and use or test versions other than the
+system-installed GStreamer version.
+
+All of these modules have now (as of 28 September 2021) been merged into
+a single git repository (“Mono repository” or “monorepo”) which should
+simplify development workflows and continuous integration, especially
+where changes need to be made to multiple modules at once.
+
+This mono repository merge will primarily affect GStreamer developers
+and contributors and anyone who has workflows based on the GStreamer git
+repositories.
+
+The Rust bindings and Rust plugins modules have not been merged into the
+mono repository at this time because they follow a different release
+cycle.
+
+The mono repository lives in the existing GStreamer core git repository
+in GitLab in the new main branch and all future development will happen
+on this branch.
+
+Modules will continue to be released as separate tarballs.
+
+For more details, please see the GStreamer mono repository FAQ.
+
+GstPlay: new high-level playback library replacing GstPlayer
+
+- GstPlay is a new high-level playback library that replaces the older
+ GstPlayer API. It is basically the same API as GstPlayer but
+ refactored to use bus messages for application notifications instead
+ of GObject signals. There is still a signal adapter object for those
+ who prefer signals. Since the existing GstPlayer API is already in
+ use in various applications, it didn’t seem like a good idea to
+ break it entirely. Instead a new API was added, and it is expected
+ that this new GstPlay API will be moved to gst-plugins-base in
+ future.
+
+- The existing GstPlayer API is scheduled for deprecation and will be
+ removed at some point in the future (e.g. in GStreamer 1.24), so
+ application developers are urged to migrate to the new GstPlay API
+ at their earliest convenience.
+
+WebM alpha decoding
+
+- Implement WebM alpha decoding (VP8/VP9 with alpha), which required
+ support and additions in various places. This is supported both with
+ software decoders and hardware-accelerated decoders.
+
+- VP8/VP9 don’t support alpha components natively in the codec, so the
+ way this is implemented in WebM is by encoding the alpha plane with
+ transparency data as a separate VP8/VP9 stream. Inside the WebM
+ container (a variant of Matroska) this is coded as a single video
+ track with the “normal” VP8/VP9 video data making up the main video
+ data and each frame of video having an encoded alpha frame attached
+ to it as extra data ("BlockAdditional").
+
+- matroskademux has been extended extract this per-frame alpha side
+ data and attach it in form of a GstVideoCodecAlphaMeta to the
+ regular video buffers. Note that this new meta is specific to this
+ VP8/VP9 alpha support and can’t be used to just add alpha support to
+ other codecs that don’t support it. Lastly, matroskademux also
+ advertises the fact that the streams contain alpha in the caps.
+
+- The new codecalpha plugin contains various bits of infrastructure to
+ support autoplugging and debugging:
+
+ - codecalphademux splits out the alpha stream from the metas on
+ the regular VP8/VP9 buffers
+ - alphacombine takes two decoded raw video streams (one alpha, one
+ the regular video) and combines it into a video stream with
+ alpha
+ - vp8alphadecodebin + vp9alphadecodebin are wrapper bins that use
+ the regular vp8dec and vp9dec software decoders to decode
+ regular and alpha streams and combine them again. To decodebin
+ these look like regular decoders which ju
+ - The V4L2 CODEC plugin has stateless VP8/VP9 decoders that can
+ decode both alpha and non-alpha stream with a single decoder
+ instance
+
+- A new AV12 video format was added which is basically NV12 with an
+ alpha plane, which is more convenient for many hardware-accelerated
+ decoders.
+
+- Watch Nicolas Dufresne’s LCA 2022 talk “Bringing WebM Alpha support
+ to GStreamer” for all the details and a demo.
+
+RTP Header Extensions Base Class and Automatic Header Extension Handling in RTP Payloaders and Depayloaders
+
+- RTP Header Extensions are specified in RFC 5285 and provide a way to
+ add small pieces of data to RTP packets in between the RTP header
+ and the RTP payload. This is often used for per-frame metadata,
+ extended timestamps or other application-specific extra data. There
+ are several commonly-used extensions specified in various RFCs, but
+ senders are free to put any kind of data in there, as long as sender
+ and receiver both know what that data is. Receivers that don’t know
+ about the header extensions will just skip the extra data without
+ ever looking at it. These header extensions can often be combined
+ with any kind of payload format, so may need to be supported by many
+ RTP payloader and depayloader elements.
+
+- Inserting and extracting RTP header extension data has so far been a
+ bit inconvenient in GStreamer: There are functions to add and
+ retrieve RTP header extension data from RTP packets, but nothing
+ works automatically, even for common extensions. People would have
+ to do the insertion/extraction either in custom elements
+ before/after the RTP payloader/depayloader, or inside pad probes,
+ which isn’t very nice.
+
+- This release adds various pieces of new infrastructure for generic
+ RTP header extension handling, as well as some implementations for
+ common extensions:
+
+ - GstRTPHeaderExtension is a new helper base class for reading and
+ writing RTP header extensions. Nominally this subclasses
+ GstElement, but only so these extensions are stored in the
+ registry where they can be looked up by URI or name. They don’t
+ have pads and don’t get added to the pipeline graph as an
+ element.
+
+ - "add-extension" and "clear-extension" action signals on RTP
+ payloaders and depayloaders for manual extension management
+
+ - The "request-extension" signal will be emitted if an extension
+ is encountered that requires explicit mapping by the application
+
+ - new "auto-header-extension" property on RTP payloaders and
+ depayloaders for automatic handling of known header extensions.
+ This is enabled by default. The extensions must be signalled via
+ caps / SDP.
+
+ - RTP header extension implementations:
+
+ - rtphdrextclientaudiolevel: Client-to-Mixer Audio Level
+ Indication (RFC 6464) (also see below)
+ - rtphdrextcolorspace: Color Space extension, extends RTP
+ packets with color space and high dynamic range (HDR)
+ information
+ - rtphdrexttwcc: Transport Wide Congestion Control support
+
+- gst_rtp_buffer_remove_extension_data() is a new helper function to
+ remove an RTP header extension from an RTP buffer
+
+- The existing gst_rtp_buffer_set_extension_data() now also supports
+ shrinking the extension data in size
+
+AppSink and AppSrc improvements
+
+- appsink: new API to pull events out of appsink in addition to
+ buffers and buffer lists.
+
+ There was previously no way for users to receive incoming events
+ from appsink properly serialised with the data flow, even if they
+ are serialised events. The reason for that is that the only way to
+ intercept events was via a pad probe on the appsink sink pad, but
+ there is also internal queuing inside of appsink, so it’s difficult
+ to ascertain the right order of everything in all cases.
+
+ There is now a new "new-serialized-event" signal which will be
+ emitted when there’s a new event pending (just like the existing
+ "new-sample" signal). The "emit-signals" property must be set to
+ TRUE in order to activate this (but it’s also fine to just pull from
+ the application thread without using the signals).
+
+ gst_app_sink_pull_object() and gst_app_sink_try_pull_object() can be
+ used to pull out either an event or a new sample carrying a buffer
+ or buffer list, whatever is next in the queue.
+
+ EOS events will be filtered and will not be returned. EOS handling
+ can be done the ususal way, same as with _pull_sample().
+
+- appsrc: allow configuration of internal queue limits in time and
+ buffers and add leaky mode.
+
+ There is internal queuing inside appsrc so the application thread
+ can push data into the element which will then be picked up by the
+ source element’s streaming thread and pushed into the pipeline from
+ that streaming thread. This queue is unlimited by default and until
+ now it was only possible to set a maximum size limit in bytes. When
+ that byte limit is reached, the pushing thread (application thread)
+ would be blocked until more space becomes available.
+
+ A limit in bytes is not particularly useful for many use cases, so
+ now it is possible to also configure limits in time and buffers
+ using the new "max-time" and "max-buffers" properties. Of course
+ there are also matching new read-only"current-level-buffers" and
+ "current-level-time properties" properties to query the current fill
+ level of the internal queue in time and buffers.
+
+ And as if that wasn’t enough the internal queue can also be
+ configured as leaky using the new "leaky-type" property. That way
+ when the queue is full the application thread won’t be blocked when
+ it tries to push in more data, but instead either the new buffer
+ will be dropped or the oldest data in the queue will be dropped.
+
+Better string serialization of nested GstCaps and GstStructures
+
+- New string serialisation format for structs and caps that can handle
+ nested structs and caps properly by using brackets to delimit nested
+ items (e.g. some-struct, some-field=[nested-struct, nested=true]).
+ Unlike the default format the new variant can also support more than
+ one level of nesting. For backwards-compatibility reasons the old
+ format is still output by default when serialising caps and structs
+ using the existing API. The new functions gst_caps_serialize() and
+ gst_structure_serialize() can be used to output strings in the new
+ format.
+
+Convenience API for custom GstMetas
+
+- New convenience API to register and create custom GstMetas:
+ gst_meta_register_custom() and gst_buffer_add_custom_meta(). Such
+ custom meta is backed by a GstStructure and does not require that
+ users of the API expose their GstMeta implementation as public API
+ for other components to make use of it. In addition, it provides a
+ simpler interface by ignoring the impl vs. api distinction that the
+ regular API exposes. This new API is meant to be the meta
+ counterpart to custom events and messages, and to be more convenient
+ than the lower-level API when the absolute best performance isn’t a
+ requirement. The reason it’s less performant than a “proper” meta is
+ that a proper meta is just a C struct in the end whereas this goes
+ through the GstStructure API which has a bit more overhead, which
+ for most scenarios is negligible however. This new API is useful for
+ experimentation or proprietary metas, but also has some limitations:
+ it can only be used if there’s a single producer of these metas;
+ it’s not allowed to register the same custom meta multiple times or
+ from multiple places.
+
+Additional Element Properties on Encoding Profiles
+
+- GstEncodingProfile: The new "element-properties" and
+ gst_encoding_profile_set_element_properties() API allows
+ applications to set additional element properties on encoding
+ profiles to configure muxers and encoders. So far the encoding
+ profile template was the only place where this could be specified,
+ but often what applications want to do is take a ready-made encoding
+ profile shipped by GStreamer or the application and then tweak the
+ settings on top of that, which is now possible with this API. Since
+ applications can’t always know in advance what encoder element will
+ be used in the end, it’s even possible to specify properties on a
+ per-element basis.
+
+ Encoding Profiles are used in the encodebin, transcodebin and
+ camerabin elements and APIs to configure output formats (containers
+ and elementary streams).
+
+Audio Level Indication Meta for RFC 6464
+
+- New GstAudioLevelMeta containing Audio Level Indication as per RFC
+ 6464
+
+- The level element has been updated to add GstAudioLevelMeta on
+ buffers if the "audio-level-meta" property is set to TRUE. This can
+ then in turn be picked up by RTP payloaders to signal the audio
+ level to receivers through RTP header extensions (see above).
+
+- New Client-to-Mixer Audio Level Indication (RFC6464) RTP Header
+ Extension which should be automatically created and used by RTP
+ payloaders and depayloaders if their "auto-header-extension"
+ property is enabled and if the extension is part of the RTP caps.
+
+Automatic packet loss, data corruption and keyframe request handling for video decoders
+
+- The GstVideoDecoder base class has gained various new APIs to
+ automatically handle packet loss and data corruption better by
+ default, especially in RTP, RTSP and WebRTC streaming scenarios, and
+ to give subclasses more control about how they want to handle
+ missing data:
+
+ - Video decoder subclasses can mark output frames as corrupted via
+ the new GST_VIDEO_CODEC_FRAME_FLAG_CORRUPTED flag
+
+ - A new "discard-corrupted-frames" property allows applications to
+ configure decoders so that corrupted frames are directly
+ discarded instead of being forwarded inside the pipeline. This
+ is a replacement for the "output-corrupt" property of the FFmpeg
+ decoders.
+
+ - RTP depayloaders can now signal to decoders that data is missing
+ when sending GAP events for lost packets. GAP events can be sent
+ for various reason in a GStreamer pipeline. Often they are just
+ used to let downstream elements know that there isn’t a buffer
+ available at the moment, so downstream elements can move on
+ instead of waiting for one. They are also sent by RTP
+ depayloaders in the case that packets are missing, however, and
+ so far a decoder was not able to differentiate the two cases.
+ This has been remedied now: GAP events can be decorated with
+ gst_event_set_gap_flags() and GST_GAP_FLAG_MISSING_DATA to let
+ decoders now what happened, and decoders can then use that in
+ some cases to handle missing data better.
+
+ - The GstVideoDecoder::handle_missing_data vfunc was added to
+ inform subclasses about packet loss or missing data and let them
+ handle it in their own way if they like.
+
+ - gst_video_decoder_set_needs_sync_point() lets subclasses signal
+ that they need the stream to start with a sync point. If
+ enabled, the base class will discard all non-sync point frames
+ in the beginning and after a flush and does not pass them to the
+ subclass. Furthermore, if the first frame is not a sync point,
+ the base class will try and request a sync frame from upstream
+ by sending a force-key-unit event (see next items).
+
+ - New "automatic-request-sync-points" and
+ "automatic-request-sync-point-flags" properties to automatically
+ request sync points when needed, e.g. on packet loss or if the
+ first frame is not a keyframe. Applications may want to enable
+ this on decoders operating in e.g. RTP/WebRTC/RTSP receiver
+ pipelines.
+
+ - The new "min-force-key-unit-interval" property can be used to
+ ensure there’s a minimal interval between keyframe requests to
+ upstream (and/or the sender) and we’re not flooding the sender
+ with key unit requests.
+
+ - gst_video_decoder_request_sync_point() allows subclasses to
+ request a new sync point (e.g. if they choose to do their own
+ missing data handling). This will still honour the
+ "min-force-key-unit-interval" property if set.
+
+Improved support for custom minimal GStreamer builds
+
+- Element registration and registration of other plugin features
+ inside plugin init functions has been improved in order to
+ facilitate minimal custom GStreamer builds.
+
+- A number of new macros have been added to declare and create
+ per-element and per-pluginfeature register functions in all plugins,
+ and then call those from the per-plugin plugin_init functions:
+
+ - GST_ELEMENT_REGISTER_DEFINE,
+ GST_DEVICE_PROVIDER_REGISTER_DEFINE,
+ GST_DYNAMIC_TYPE_REGISTER_DEFINE, GST_TYPE_FIND_REGISTER_DEFINE
+ for the actual registration call with GStreamer
+ - GST_ELEMENT_REGISTER, GST_DEVICE_PROVIDER_REGISTER,
+ GST_DYNAMIC_TYPE_REGISTER, GST_PLUGIN_STATIC_REGISTER,
+ GST_TYPE_FIND_REGISTER to call the registration function defined
+ by the REGISTER_DEFINE macro
+ - GST_ELEMENT_REGISTER_DECLARE,
+ GST_DEVICE_PROVIDER_REGISTER_DECLARE,
+ GST_DYNAMIC_TYPE_REGISTER_DECLARE,
+ GST_TYPE_FIND_REGISTER_DECLARE to declare the registration
+ function defined by the REGISTER_DEFINE macro
+ - and various variants for advanced use cases.
+
+- This means that applications can call the per-element and
+ per-pluginfeature registration functions for only the elements they
+ need instead of registering plugins as a whole with all kinds of
+ elements that may not be required (e.g. encoder and decoder instead
+ of just decoder). In case of static linking all unused functions and
+ their dependencies would be removed in this case by the linker,
+ which helps minimise binary size for custom builds.
+
+- gst_init() will automatically call a gst_init_static_plugins()
+ function if one exists.
+
+- See the GStreamer static build documentation and Stéphane’s blog
+ post Generate a minimal GStreamer build, tailored to your needs for
+ more details.
New elements
-- this section will be filled in in due course
+- New aesdec and aesenc elements for AES encryption and decryption in
+ a custom format.
+
+- New encodebin2 element with dynamic/sometimes source pads in order
+ to support the option of doing the muxing outside of encodebin,
+ e.g. in combination with a splitmuxsink.
+
+- New fakeaudiosink and videocodectestsink elements for testing and
+ debugging (see below for more details)
+
+- rtpisacpay, rtpisacdepay: new RTP payloader and depayloader for iSAC
+ audio codec
+
+- rtpst2022-1-fecdec, rtpst2022-1-fecenc: new elements providing SMPTE
+ 2022-1 2-D Forward Error Correction. More details in Mathieu’s blog
+ post.
+
+- isac: new plugin wrapping the Internet Speech Audio Codec reference
+ encoder and decoder from the WebRTC project.
+
+- asio: plugin for Steinberg ASIO (Audio Streaming Input/Output) API
+
+- gssrc, gssink: add source and sink for Google Cloud Storage
+
+- onnx: new plugin to apply ONNX neural network models to video
+
+- openaptx: aptX and aptX-HD codecs using libopenaptx (v0.2.0)
+
+- qroverlay, debugqroverlay: new elements that allows overlaying data
+ on top of video in form of a QR code
+
+- cvtracker: new OpenCV-based tracker element
+
+- av1parse, vp9parse: new parsers for AV1 and VP9 video
+
+- va: work on the new VA-API plugin implementation for
+ hardware-accelerated video decoding and encoding has continued at
+ pace, with various new decoders and filters having joined the
+ initial vah264dec:
+
+ - vah265dec: VA-API H.265 decoder
+ - vavp8dec: VA-API VP8 decoder
+ - vavp9dec: VA-API VP9 decoder
+ - vaav1dec: VA-API AV1 decoder
+ - vampeg2dec: VA-API MPEG-2 decoder
+ - vadeinterlace: : VA-API deinterlace filter
+ - vapostproc: : VA-API postproc filter (color conversion,
+ resizing, cropping, color balance, video rotation, skin tone
+ enhancement, denoise, sharpen)
+
+ See Víctor’s blog post “GstVA in GStreamer 1.20” for more details
+ and what’s coming up next.
+
+- vaapiav1dec: new AV1 decoder element (in gstreamer-vaapi)
+
+- msdkav1dec: hardware-accelerated AV1 decoder using the Intel Media
+ SDK / oneVPL
+
+- nvcodec plugin for NVIDIA NVCODEC API for hardware-accelerated video
+ encoding and decoding:
+
+ - cudaconvert, cudascale: new CUDA based video color space convert
+ and rescale elements
+ - cudaupload, cudadownload: new helper elements for memory
+ transfer between CUDA and system memory spaces
+ - nvvp8sldec, nvvp9sldec: new GstCodecs-based VP8/VP9 decoders
+
+- Various new hardware-accelerated elements for Windows:
+
+ - d3d11screencapturesrc: new desktop capture element, including a
+ GstDeviceProvider implementation to enumerate/select target
+ monitors for capture.
+ - d3d11av1dec and d3d11mpeg2dec: AV1 and MPEG-2 decoders
+ - d3d11deinterlace: deinterlacing filter
+ - d3d11compositor: video composing element
+ - see Windows section below for more details
+
+- new Rust plugins:
+
+ - audiornnoise: Removes noise from an audio stream
+ - awstranscribeparse: Parses AWS audio transcripts into timed text
+ buffers
+ - ccdetect: Detects if valid closed captions are present in a
+ closed captions stream
+ - cea608tojson: Converts CEA-608 Closed Captions to a JSON
+ representation
+ - cmafmux: CMAF fragmented MP4 muxer
+ - dashmp4mux: DASH fragmented MP4 muxer
+ - isofmp4mux: ISO fragmented MP4 muxer
+ - ebur128level: EBU R128 Loudness Level Measurement
+ - ffv1dec: FFV1 video decoder
+ - gtk4paintablesink: GTK4 video sink, which provides a
+ GdkPaintable that can be rendered in various widgets
+ - hlssink3: HTTP Live Streaming sink
+ - hrtfrender: Head-Related Transfer Function (HRTF) renderer
+ - hsvdetector: HSV colorspace detector
+ - hsvfilter: HSV colorspace filter
+ - jsongstenc: Wraps buffers containing any valid top-level JSON
+ structures into higher level JSON objects, and outputs those as
+ ndjson
+ - jsongstparse: Parses ndjson as output by jsongstenc
+ - jsontovtt: converts JSON to WebVTT subtitles
+ - regex: Applies regular expression operations on text
+ - roundedcorners: Adds rounded corners to video
+ - spotifyaudiosrc: Spotify source
+ - textahead: Display upcoming text buffers ahead (e.g. for
+ Karaoke)
+ - transcriberbin: passthrough bin that transcribes raw audio to
+ closed captions using awstranscriber and puts the captions as
+ metas onto the video
+ - tttojson: Converts timed text to a JSON representation
+ - uriplaylistbin: Playlist source bin
+ - webpdec-rs: WebP image decoder with animation support
+
+- New plugin codecalpha with elements to assist with WebM Alpha
+ decoding
+
+ - codecalphademux: Split stream with GstVideoCodecAlphaMeta into
+ two streams
+ - alphacombine: Combine two raw video stream (I420 or NV12) as one
+ stream with alpha channel (A420 or AV12)
+ - vp8alphadecodebin: A bin to handle software decoding of VP8 with
+ alpha
+ - vp9alphadecodebin: A bin to handle software decoding of VP9 with
+ alpha
+
+- New hardware accelerated elements for Linux:
+
+ - v4l2slmpeg2dec: Support for Linux Stateless MPEG2 decoders
+ - v4l2slvp9dec: Support for Linux Stateless VP9 decoders
+ - v4l2slvp8alphadecodebin: Support HW accelerated VP8 with alpha
+ layer decoding
+ - v4l2slvp9alphadecodebin: Support HW accelerated VP9 with alpha
+ layer decoding
New element features and additions
-- this section will be filled in in due course
+- assrender: handle more font mime types; better interaction with
+ matroskademux for embedded fonts
+
+- audiobuffersplit: Add support for specifying output buffer size in
+ bytes (not just duration)
+
+- audiolatency: new "samplesperbuffer" property so users can configure
+ the number of samples per buffer. The default value is 240 samples
+ which is equivalent to 5ms latency with a sample rate of 48000,
+ which might be larger than actual buffer size of audio capture
+ device.
+
+- audiomixer, audiointerleave, GstAudioAggregator: now keep a count of
+ samples that are dropped or processed as statistic and can be made
+ to post QoS messages on the bus whenever samples are dropped by
+ setting the "qos-messages" property on input pads.
+
+- audiomixer, compositor: improved handling of new inputs added at
+ runtime. New API was added to the GstAggregator base class to allow
+ subclasses to opt into an aggregation mode where inactive pads are
+ ignored when processing input buffers
+ (gst_aggregator_set_ignore_inactive_pads(),
+ gst_aggregator_pad_is_inactive()). An “inactive pad” in this context
+ is a pad which, in live mode, hasn’t yet received a first buffer,
+ but has been waited on at least once. What would happen usually in
+ this case is that the aggregator would wait for data on this pad
+ every time, up to the maximum configured latency. This would
+ inadvertently push mixer elements in live mode to the configured
+ latency envelope and delay processing when new inputs are added at
+ runtime until these inputs have actually produced data. This is
+ usually undesirable. With this new API, new inputs can be added
+ (requested) and configured and they won’t delay the data processing.
+ Applications can opt into this new behaviour by setting the
+ "ignore-inactive-pads" property on compositor, audiomixer or other
+ GstAudioAggregator-based elements.
+
+- cccombiner: implement “scheduling” of captions. So far cccombiner’s
+ behaviour was essentially that of a funnel: it strictly looked at
+ input timestamps to associate together video and caption buffers.
+ Now it will try to smoothly schedule caption buffers in order to
+ have exactly one per output video buffer. This might involve
+ rewriting input captions, for example when the input is CDP then
+ sequence counters are rewritten, time codes are dropped and
+ potentially re-injected if the input video frame had a time code
+ meta. This can also lead to the input drifting from synchronization,
+ when there isn’t enough padding in the input stream to catch up. In
+ that case the element will start dropping old caption buffers once
+ the number of buffers in its internal queue reaches a certain limit
+ (configurable via the "max-scheduled" property). The new original
+ funnel-like behaviour can be restored by setting the "scheduling"
+ property to FALSE.
+
+- ccconverter: new "cdp-mode" property to specify which sections to
+ include in CDP packets (timecode, CC data, service info). Various
+ software, including ffmpeg’s Decklink support, fails parsing CDP
+ packets that contain anything but CC data in the CDP packets.
+
+- clocksync: new "sync-to-first" property for automatic timestamp
+ offset setup: if set clocksync will set up the "ts-offset" value
+ based on the first buffer and the pipeline’s running time when the
+ first buffer arrived. The newly configured "ts-offset" in this case
+ would be the value that allows outputting the first buffer without
+ waiting on the clock. This is useful for example to feed a non-live
+ input into an already-running pipeline.
+
+- compositor:
+
+ - multi-threaded input conversion and compositing. Set the
+ "max-threads" property to activate this.
+ - new "sizing-policy" property to support display aspect ratio
+ (DAR)-aware scaling. By default the image is scaled to fill the
+ configured destination rectangle without padding and without
+ keeping the aspect ratio. With sizing-policy=keep-aspect-ratio
+ the input image is scaled to fit the destination rectangle
+ specified by GstCompositorPad:{xpos, ypos, width, height}
+ properties preserving the aspect ratio. As a result, the image
+ will be centered in the destination rectangle with padding if
+ necessary.
+ - new "zero-size-is-unscaled" property on input pads. By default
+ pad width=0 or pad height=0 mean that the stream should not be
+ scaled in that dimension. But if the "zero-size-is-unscaled"
+ property is set to FALSE a width or height of 0 is instead
+ interpreted to mean that the input image on that pad should not
+ be composited, which is useful when creating animations where an
+ input image is made smaller and smaller until it disappears.
+ - improved handling of new inputs at runtime via
+ "ignore-inactive-pads"property (see above for details)
+ - allow output format with alpha even if none of the inputs have
+ alpha (also glvideomixer and other GstVideoAggregator
+ subclasses)
+
+- dashsink: add h265 codec support and signals for allowing custom
+ playlist/fragment output
+
+- decodebin3:
+
+ - improved decoder selection, especially for hardware decoders
+ - make input activation “atomic” when adding inputs dynamically
+ - better interleave handling: take into account decoder latency
+ for interleave size
+
+- decklink:
+
+ - Updated DeckLink SDK to 11.2 to support DeckLink 8K Pro
+ - decklinkvideosrc:
+ - More accurate and stable capture timestamps: use the
+ hardware reference clock time when the frame was finished
+ being captured instead of a clock time much further down the
+ road.
+ - Automatically detect widescreen vs. normal NTSC/PAL
+
+- encodebin:
+
+ - add “smart encoding” support for H.265, VP8 and VP9 (i.e. only
+ re-encode where needed and otherwise pass through encoded video
+ as-is).
+ - H264/H265 smart encoding improvements: respect user-specified
+ stream-format, but if not specified default to avc3/hvc1 with
+ in-band SPS/PPS/VPS signalling for more flexibility.
+ - new encodebin2 element with dynamic/sometimes source pads in
+ order to support the option of doing the muxing outside of
+ encodebin, e.g. in combination with splitmuxsink.
+ - add APIs to set element properties on encoding profiles (see
+ below)
+
+- errorignore: new "ignore-eos" property to also ignore FLOW_EOS from
+ downstream elements
+
+- giosrc: add support for growing source files: applications can
+ specify that the underlying file being read is growing by setting
+ the "is-growing" property. If set, the source won’t EOS when it
+ reaches the end of the file, but will instead start monitoring it
+ and will start reading data again whenever a change is detected. The
+ new "waiting-data" and "done-waiting-data" signals keep the
+ application informed about the current state.
+
+- gtksink, gtkglsink:
+
+ - scroll event support: forwarded as navigation events into the
+ pipeline
+ - "video-aspect-ratio-override" property to force a specific
+ aspect ratio
+ - "rotate-method" property and support automatic rotation based on
+ image tags
+
+- identity: new "stats" property allows applications to retrieve the
+ number of bytes and buffers that have passed through so far.
+
+- interlace: add support for more formats, esp 10-bit, 12-bit and
+ 16-bit ones
+
+- jack: new "low-latency" property for automatic latency-optimized
+ setting and "port-names" property to select ports explicitly
+
+- jpegdec: support output conversion to RGB using libjpeg-turbo (for
+ certain input files)
+
+- line21dec:
+
+ - "mode" property to control whether and how detected closed
+ captions should be inserted in the list of existing close
+ caption metas on the input frame (if any): add, drop, or
+ replace.
+ - "ntsc-only" property to only look for captions if video has NTSC
+ resolution
+
+- line21enc: new "remove-caption-meta" to remove metas from output
+ buffers after encoding the captions into the video data; support for
+ CDP closed captions
+
+- matroskademux, matroskamux: Add support for ffv1, a lossless
+ intra-frame video coding format.
+
+- matroskamux: accept in-band SPS/PPS/VPS for H264 and H265
+ (i.e. stream-format avc3 and hev1) which allows on-the-fly
+ profile/level/resolution changes.
+
+- matroskamux: new "cluster-timestamp-offset" property, useful for use
+ cases where the container timestamps should map to some absolute
+ wall clock time, for example.
+
+- rtpsrc: add "caps" property to allow explicit setting of the caps
+ where needed
+
+- mpegts: support SCTE-35 passthrough via new "send-scte35-events"
+ property on MPEG-TS demuxer tsdemux. When enabled, SCTE 35 sections
+ (eg ad placement opportunities) are forwarded as events donwstream
+ where they can be picked up again by mpegtsmux. This required a
+ semantic change in the SCTE-35 section API: timestamps are now in
+ running time instead of muxer pts.
+
+- tsdemux: Handle PCR-less MPEG-TS streams; more robust timestamp
+ handling in certain corner cases and for poorly muxed streams.
+
+- mpegtsmux:
+
+ - More conformance improvements to make MPEG-TS analyzers happy:
+ - PCR timing accuracy: Improvements to the way mpegtsmux
+ outputs PCR observations in CBR mode, so that a PCR
+ observation is always inserted when needed, so that we never
+ miss the configured pcr-interval, as that triggers various
+ MPEG-TS analyser errors.
+ - Improved PCR/SI scheduling
+ - Don’t write PCR until PAT/PMT are output to make sure streams
+ start cleanly with a PAT/PMT.
+ - Allow overriding the automatic PMT PID selection via
+ application-supplied PMT_%d fields in the prog-map
+ structure/property.
+
+- mp4mux:
+
+ - new "first-moov-then-finalise" mode for fragmented output where
+ the output will start with a self-contained moov atom for the
+ first fragment, and then produce regular fragments. Then at the
+ end when the file is finalised, the initial moov is invalidated
+ and a new moov is written covering the entire file. This way the
+ file is a “fragmented mp4” file while it is still being written
+ out, and remains playable at all times, but at the end it is
+ turned into a regular mp4 file (with former fragment headers
+ remaining as unused junk data in the file).
+ - support H.264 avc3 and H.265 hvc1 stream formats as input where
+ the codec data is signalled in-band inside the bitstream instead
+ of caps/file headers.
+ - support profile/level/resolution changes for H264/H265 input
+ streams (i.e. codec data changing on the fly). Each codec_data
+ is put into its own SampleTableEntry inside the stsd, unless the
+ input is in avc3 stream format in which case it’s written
+ in-band an not in the headers.
+
+- multifilesink: new ""min-keyframe-distance"" property to make
+ minimum distance between keyframes in next-file=key-frame mode
+ configurable instead of hard-coding it to 10 seconds.
+
+- mxfdemux has seen a big refactoring to support non-frame wrappings
+ and more accurate timestamp/seek handling for some formats
+
+- msdk plugin for hardware-accelerated video encoding and decoding
+ using the Intel Media SDK:
+
+ - oneVPL support (Intel oneAPI Video Processing Library)
+ - AV1 decoding support
+ - H.264 decoder now supports constrained-high and progressive-high
+ profiles
+ - H.264 encoder:
+ - more configuration options (properties):
+ "intra-refresh-type", "min-qp" , "max-qp", "p-pyramid",
+ "dblk-idc"
+ - H.265 encoder:
+ - can output main-still-picture profile
+ - now inserts HDR SEIs (mastering display colour volume and
+ content light level)
+ - more configuration options (properties):
+ "intra-refresh-type", "min-qp" , "max-qp", "p-pyramid",
+ "b-pyramid", "dblk-idc", "transform-skip"
+ - support for RGB 10bit format
+ - External bitrate control in encoders
+ - Video post proc element msdkvpp gained support for 12-bit pixel
+ formats P012_LE, Y212_LE and Y412_LE
+
+- nvh264sldec: interlaced stream support
+
+- openh264enc: support main, high, constrained-high and
+ progressive-high profiles
+
+- openjpeg: support for multithreaded decoding and encoding
+
+- rtspsrc: now supports IPv6 also for tunneled mode (RTSP-over-HTTP);
+ new "ignore-x-server-reply" property to ignore the
+ x-server-ip-address server header reply in case of HTTP tunneling,
+ as it is often broken.
+
+- souphttpsrc: Runtime compatibility support for libsoup2 and
+ libsoup3. libsoup3 is the latest major version of libsoup, but
+ libsoup2 and libsoup3 can’t co-exist in the same process because
+ there is no namespacing or versioning for GObject types. As a
+ result, it would be awkward if the GStreamer souphttpsrc plugin
+ linked to a specific version of libsoup, because it would only work
+ with applications that use the same version of libsoup. To make this
+ work, the soup plugin now tries to determine the libsoup version
+ used by the application (and its other dependencies) at runtime on
+ systems where GStreamer is linked dynamically. libsoup3 support is
+ still considered somewhat experimental at this point.
+
+- srtsrc, srtsink: add signals for the application to accept/reject
+ incoming connections
+
+- timeoverlay: new elapsed-running-time time mode which shows the
+ running time since the first running time (and each flush-stop).
+
+- udpsrc: new timestamping mode to retrieve packet receive timestamps
+ from the kernel via socket control messages (SO_TIMESTAMPNS) on
+ supported platforms
+
+- uritranscodebin: new setup-source and element-setup signals for
+ applications to configure elements used
+
+- v4l2codecs plugin gained support for 4x4 and 32x32 tile formats
+ enabling some platforms or direct renders. Important memory usage
+ improvement.
+
+- v4l2slh264dec now implements the final Linux uAPI as shipped on
+ Linux 5.11 and later.
+
+- valve: add "drop-mode" property and provide two new modes of
+ operation: in drop-mode=forward-sticky-events sticky events
+ (stream-start, segment, tags, caps, etc.) are forwarded downstream
+ even when dropping is enabled; drop-mode=transform-to-gap will in
+ addition also convert buffers into gap events when dropping is
+ enabled, which lets downstream elements know that time is advancing
+ and might allow for preroll in many scenarios. By default all events
+ and all buffers are dropped when dropping is enabled, which can
+ cause problems with caps negotiation not progressing or branches not
+ prerolling when dropping is enabled.
+
+- videocrop: support for many more pixel formats, e.g. planar YUV
+ formats with > 8bits and GBR* video formats; can now also accept
+ video not backed by system memory as long as downstream supports the
+ GstCropMeta
+
+- videotestsrc: new smpte-rp-219 pattern for SMPTE75 RP-219 conformant
+ color bars
+
+- vp8enc: finish support for temporal scalability: two new properties
+ ("temporal-scalability-layer-flags",
+ "temporal-scalability-layer-sync-flags") and a unit change on the
+ "temporal-scalability-target-bitrate" property (now expects bps);
+ also make temporal scalability details available to RTP payloaders
+ as buffer metadata.
+
+- vp9enc: new properties to tweak encoder performance:
+
+ - "aq-mode" to configure adaptive quantization modes
+ - "frame-parallel-decoding" to configure whether to create a
+ bitstream that reduces decoding dependencies between frames
+ which allows staged parallel processing of more than one video
+ frames in the decoder. (Defaults to TRUE)
+ - "row-mt", "tile-columns" and "tile-rows" so multithreading can
+ be enabled on a per-tile basis, instead of on a per tile-column
+ basis. In combination with the new "tile-rows" property, this
+ allows the encoder to make much better use of the available CPU
+ power.
+
+- vp9dec, vp9enc: add support for 10-bit 4:2:0 and 4:2:2 YUV, as well
+ as 8-bit 4:4:4
+
+- vp8enc, vp9enc now default to “good quality” for the deadline
+ property rather then “best quality”. Having the deadline set to best
+ quality causes the encoder to be absurdly slow, most real-life users
+ will prefer good-enough quality with better performance instead.
+
+- wpesrc:
+
+ - implement audio support: a new sometimes source pad will be
+ created for each audio stream created by the web engine.
+ - move wpesrc to wpevideosrc and add a wrapper bin wpesrc to also
+ support audio
+ - also handles web:// URIs now (same as cefsrc)
+ - post messages with the estimated load progress on the bus
-Plugin and library moves
+- x265enc: add negative DTS support, which means timestamps are now
+ offset by 1h same as with x264enc
+
+RTP Payloaders and Depayloaders
+
+- rtpisacpay, rtpisacdepay: new RTP payloader and depayloader for iSAC
+ audio codec
+
+- rtph264depay:
+
+ - new "request-keyframe" property to make the depayloader
+ automatically request a new keyframe from the sender on packet
+ loss, consistent with the new property on rtpvp8depay.
+ - new "wait-for-keyframe" property to make depayloader wait for a
+ new keyframe at the beginning and after packet loss (only
+ effective if the depayloader outputs AUs), consistent with the
+ existing property on rtpvp8depay.
+
+- rtpopuspay, rtpopusdepay: support libwebrtc-compatible multichannel
+ audio in addition to the previously supported multichannel audio
+ modes
+
+- rtpopuspay: add DTX (Discontinuous Transmission) support
+
+- rtpvp8depay: new "request-keyframe" property to make the depayloader
+ automatically request a new keyframe from the sender on packet loss.
+
+- rtpvp8pay: temporal scaling support
-- this section will be filled in in due course
+- rtpvp9depay: Improved SVC handling (aggregate all layers)
+
+RTP Infrastructure
+
+- rtpst2022-1-fecdec, rtpst2022-1-fecenc: new elements providing SMPTE
+ 2022-1 2-D Forward Error Correction. More details in Mathieu’s blog
+ post.
+
+- rtpreddec: BUNDLE support
+
+- rtpredenc, rtpulpfecenc: add support for Transport-wide Congestion
+ Control (TWCC)
+
+- rtpsession: new "twcc-feedback-interval" property to allow RTCP TWCC
+ reports to be scheduled on a timer instead of per marker-bit.
+
+Plugin and library moves
- There were no plugin moves or library moves in this cycle.
The following elements or plugins have been removed:
-- this section will be filled in in due course
-
- The ofa audio fingerprinting plugin has been removed. The MusicIP
database has been defunct for years so this plugin is likely neither
useful nor used by anyone.
Miscellaneous API additions
-- this section will be filled in in due course
+Core
+
+- gst_buffer_new_memdup() is a convenience function for the
+ widely-used gst_buffer_new_wrapped(g_memdup(data,size),size)
+ pattern.
+
+- gst_caps_features_new_single() creates a new single GstCapsFeatures,
+ avoiding the need to use the vararg function with NULL terminator
+ for simple cases.
+
+- gst_element_type_set_skip_documentation() can be used by plugins to
+ signal that certain elements should not be included in the GStreamer
+ plugin documentation. This is useful for plugins where elements are
+ registered dynamically based on hardware capabilities and/or where
+ the available plugins and properties vary from system to system.
+ This is used in the d3d11 plugin for example to ensure that only the
+ list of default elements is advertised in the documentation.
+
+- gst_type_find_suggest_empty_simple() is a new convenience function
+ for typefinders for cases where there’s only a media type and no
+ other fields.
+
+- New API to create elements and set properties at construction time,
+ which is not only convenient, but also allows GStreamer elements to
+ have construct-only properties: gst_element_factory_make_full(),
+ gst_element_factory_make_valist(),
+ gst_element_factory_make_with_properties(),
+ gst_element_factory_create_full(),
+ gst_element_factory_create_valist(),
+ gst_element_factory_create_with_properties().
+
+- GstSharedTaskPool: new “shared” task pool subclass with slightly
+ different default behaviour than the existing GstTaskPool which
+ would create unlimited number of threads for new tasks. The shared
+ taskpool creates up to N threads (default: 1) and then distributes
+ pending tasks to those threads round-robin style, and blocks if no
+ thread is available. It is possible to join tasks. This can be used
+ by plugins to implement simple multi-threaded processing and is used
+ for the new multi-threaded video conversion and compositing done in
+ GstVideoAggregator, videoconverter and compositor.
+
+Plugins Base Utils library
+
+- GstDiscoverer:
+
+ - gst_discoverer_container_info_get_tags() was added to retrieve
+ global/container tags (vs. per-stream tags). Per-Stream tags can
+ be retrieved via the existing
+ gst_discoverer_stream_info_get_tags().
+ gst_discoverer_info_get_tags(), which for many files returns a
+ confusing mix of stream and container tags, has been deprecated
+ in favour of the container/stream-specific functions.
+ - gst_discoverer_stream_info_get_stream_number() returns a unique
+ integer identifier for a given stream within the given
+ GstDiscoverer context. (If this matches the stream number inside
+ the container bitstream that’s by coincidence and not by
+ design.)
+
+- gst_pb_utils_get_caps_description_flags() can be used to query
+ whether certain caps represent a container, audio, video, image,
+ subtitles, tags, or something else. This only works for formats
+ known to GStreamer.
+
+- gst_pb_utils_get_file_extension_from_caps() returns a possible file
+ extension for given caps.
+
+- gst_codec_utils_h264_get_profile_flags_level(): Parses profile,
+ flags, and level from H264 AvcC codec_data. The format of H264 AVCC
+ extradata/sequence_header is documented in the ITU-T H.264
+ specification section 7.3.2.1.1 as well as in ISO/IEC 14496-15
+ section 5.3.3.1.2.
+
+- gst_codec_utils_caps_get_mime_codec() to convert caps to a RFC 6381
+ compatible MIME codec string codec. Useful for providing the codecs
+ field inside the Content-Type HTTP header for containerized formats,
+ such as mp4 or matroska.
+
+GStreamer OpenGL integration library and plugins
+
+- glcolorconvert: added suppport for converting the video formats
+ A420, AV12, BGR, BGRA, RGBP and BGRP.
+
+- Added support to GstGLBuffer for persistent buffer mappings where a
+ Pixel Buffer Object (PBO) can be mapped by both the CPU and the GPU.
+ This removes a memcpy() when uploading textures or vertices
+ particularly when software decoders (e.g. libav) are direct
+ rendering into our memory. Improves transfer performance
+ significantly. Requires OpenGL 4.4, GL_ARB_buffer_storage or
+ GL_EXT_buffer_storage
+
+- Added various helper functions for handling 4x4 matrices of affine
+ transformations as used by GstVideoAffineTransformationMeta.
+
+- Add support to GstGLContext for allowing the application to control
+ the config (EGLConfig, GLXConfig, etc) used when creating the OpenGL
+ context. This allows the ability to choose between RGB16 or RGB10A2
+ or RGBA8 back/front buffer configurations that were previously
+ hardcoded. GstGLContext also supports retrieving the configuration
+ it was created with or from an externally provide OpenGL context
+ handle. This infrastructure is also used to create a compatible
+ config from an application/externally provided OpenGL context in
+ order to improve compatibility with other OpenGL frameworks and GUI
+ toolkits. A new environment variable GST_GL_CONFIG was also added to
+ be able to request a specific configuration from the command line.
+ Note: different platforms will have different functionality
+ available.
+
+- Add support for choosing between EGL and WGL at runtime when running
+ on Windows. Previously this was a build-time switch. Allows use in
+ e.g. Gtk applications on Windows that target EGL/ANGLE without
+ recompiling GStreamer. gst_gl_display_new_with_type() can be used by
+ applications to choose a specific display type to use.
+
+- Build fixes to explicitly check for Broadcom-specific libraries on
+ older versions of the Raspberry Pi platform. The Broadcom OpenGL ES
+ and EGL libraries have different filenames. Using the vc4 Mesa
+ driver on the Raspberry Pi is not affected.
+
+- Added support to glupload and gldownload for transferring RGBA
+ buffers using the memory:NVMM available on the Nvidia Tegra family
+ of embedded devices.
+
+- Added support for choosing libOpenGL and libGLX as used in a GLVND
+ environment on unix-based platforms. This allows using desktop
+ OpenGL and EGL without pulling in any GLX symbols as would be
+ required with libGL.
+
+Video library
+
+- New raw video formats:
+
+ - AV12 (NV12 with alpha plane)
+ - RGBP and BGRP (planar RGB formats)
+ - ARGB64 variants with specified endianness instead of host
+ endianness:
+ - ARGB64_LE, ARGB64_BE
+ - RGBA64_BE, RGBA64_LE
+ - BGRA64_BE, BGRA64_LE
+ - ABGR64_BE, ABGR64_LE
+
+- gst_video_orientation_from_tag() is new convenience API to parse the
+ image orientation from a GstTagList.
+
+- GstVideoDecoder subframe support (see below)
+
+- GstVideoCodecState now also carries some HDR metadata
+
+- Ancillary video data: implement transform functions for AFD/Bar
+ metas, so they will be forwarded in more cases
+
+MPEG-TS library
+
+This library only handles section parsing and such, see above for
+changes to the actual mpegtsmux and mpegtsdemux elements.
+
+- many additions and improvements to SCTE-35 section parsing
+- new API for fetching extended descriptors:
+ gst_mpegts_find_descriptor_with_extension()
+- add support for SIT sections (Selection Information Tables)
+- expose event-from-section constructor gst_event_new_mpegts_section()
+- parse Audio Preselection Descriptor needed for Dolby AC-4
+
+GstWebRTC library + webrtcbin
+
+- Change the way in which sink pads and transceivers are matched
+ together to support easier usage. If a pad is created without a
+ specific index (i.e. using sink_%u as the pad template), then an
+ available compatible transceiver will be searched for. If a specific
+ index is requested (i.e. sink_1) then if a transceiver for that
+ m-line already exists, that transceiver must match the new sink pad
+ request. If there is no transceiver available in either scenario, a
+ new transceiver is created. If a mixture of both sink_1 and sink_%u
+ requests result in an impossible situation, an error will be
+ produced at pad request time or from create offer/answer.
+
+- webrtcbin now uses regular ICE nomination instead of libnice’s
+ default of aggressive ICE nomination. Regular ICE nomination is the
+ default recommended by various relevant standards and improves
+ connectivity in specific network scenarios.
+
+- Add support for limiting the port range used for RTP with the
+ addition of the min-rtp-port and max-rtp-port properties on the ICE
+ object.
+
+- Expose the SCTP transport as a property on webrtcbin to more closely
+ match the WebRTC specification.
+
+- Added support for taking into account the data channel transport
+ state when determining the value of the "connection-state" property.
+ Previous versions of the WebRTC spec did not include the data
+ channel state when computing this value.
+
+- Add configuration for choosing the size of the underlying sockets
+ used for transporting media data
+
+- Always advertise support for the transport-cc RTCP feedback protocol
+ as rtpbin supports it. For full support, the configured caps (input
+ or through codec-preferences) need to include the relevant RTP
+ header extension.
+
+- Numerous fixes to caps and media handling to fail-fast when an
+ incompatible situation is detected.
+
+- Improved support for attaching the required media after a remote
+ offer has been set.
+
+- Add support for dynamically changing the amount of FEC used for a
+ particular stream.
+
+- webrtcbin now stops further SDP processing at the first error it
+ encounters.
+
+- Completed support for either local or the remote closing a data
+ channel.
+
+- Various fixes when performing BUNDLEing of the media streams in
+ relation to RTX and FEC usage.
+
+- Add support for writing out QoS DSCP marking on outgoing packets to
+ improve reliability in some network scenarios.
+
+- Improvements to the statistics returned by the get-stats signal
+ including the addition of the raw statistics from the internal
+ RTPSource, the TWCC stats when available.
+
+- The webrtc library does not expose any objects anymore with public
+ fields. Instead properties have been added to replace that
+ functionality. If you are accessing such fields in your application,
+ switch to the corresponding properties.
+
+GstCodecs and Video Parsers
+
+- Support for render delays to improve throughput across all CODECs
+ (used with NVDEC and V4L2).
+- lots of improvements to parsers and the codec parsing decoder base
+ classes (H264, H265, VP8, VP9, AV1, MPEG-2) used for various
+ hardware-accelerate decoder APIs.
+
+Bindings support
+
+- gst_allocation_params_new() allocates a GstAllocationParams struct
+ on the heap. This should only be used by bindings (and freed via
+ gst_allocation_params_free() then). In C code you would allocate
+ this on the stack and only init it in place.
+
+- gst_debug_log_literal() can be used to log a string to the debug log
+ without going through any printf format expansion and associated
+ overhead. This is mostly useful for bindings such as the Rust
+ bindings which may have done their own formatting already .
+
+- Provide non-inlined versions of refcounting APIs for various
+ GStreamer mini objects, so that they can be consumed by bindings
+ (e.g. gstreamer-sharp): gst_buffer_ref, gst_buffer_unref,
+ gst_clear_buffer, gst_buffer_copy, gst_buffer_replace,
+ gst_buffer_list_ref, gst_buffer_list_unref, gst_clear_buffer_list,
+ gst_buffer_list_copy, gst_buffer_list_replace, gst_buffer_list_take,
+ gst_caps_ref, gst_caps_unref, gst_clear_caps, gst_caps_replace,
+ gst_caps_take, gst_context_ref, gst_context_unref, gst_context_copy,
+ gst_context_replace, gst_event_replace, gst_event_steal,
+ gst_event_take, gst_event_ref, gst_event_unref, gst_clear_event,
+ gst_event_copy, gst_memory_ref, gst_memory_unref, gst_message_ref,
+ gst_message_unref, gst_clear_message, gst_message_copy,
+ gst_message_replace, gst_message_take, gst_promise_ref,
+ gst_promise_unref, gst_query_ref, gst_query_unref, gst_clear_query,
+ gst_query_copy, gst_query_replace, gst_query_take, gst_sample_ref,
+ gst_sample_unref, gst_sample_copy, gst_tag_list_ref,
+ gst_tag_list_unref, gst_clear_tag_list, gst_tag_list_replace,
+ gst_tag_list_take, gst_uri_copy, gst_uri_ref, gst_uri_unref,
+ gst_clear_uri.
+
+- expose a GType for GstMiniObject
+
+- gst_device_provider_probe() now returns non-floating device object
+
+API Deprecations
+
+- gst_element_get_request_pad() has been deprecated in favour of the
+ newly-added gst_element_request_pad_simple() which does the exact
+ same thing but has a less confusing name that hopefully makes clear
+ that the function request a new pad rather than just retrieves an
+ already-existing request pad.
+
+- gst_discoverer_info_get_tags(), which for many files returns a
+ confusing mix of stream and container tags, has been deprecated in
+ favour of the container-specific and stream-specific functions,
+ gst_discoverer_container_info_get_tags() and
+ gst_discoverer_stream_info_get_tags().
+
+- gst_video_sink_center_rect() was deprecated in favour of the more
+ generic newly-added gst_video_center_rect().
+
+- The GST_MEMORY_FLAG_NO_SHARE flag has been deprecated, as it tends
+ to cause problems and prevents sub-buffering. If pooling or lifetime
+ tracking is required, memories should be allocated through a custom
+ GstAllocator instead of relying on the lifetime of the buffers the
+ memories were originally attached to, which is fragile anyway.
+
+- The GstPlayer high-level playback library is being replaced with the
+ new GstPlay library (see above). GstPlayer should be considered
+ deprecated at this point and will be marked as such in the next
+ development cycle. Applications should be ported to GstPlay.
+
+- Gstreamer Editing Services: ges_video_transition_set_border(),
+ ges_video_transition_get_border()
+ ges_video_transition_set_inverted()
+ ges_video_transition_is_inverted() have been deprecated, use
+ ges_timeline_element_set_children_properties() instead.
Miscellaneous performance, latency and memory optimisations
-- this section will be filled in in due course
+More video conversion fast paths
-Miscellaneous other changes and enhancements
+- v210 ↔ I420, YV12, Y42B, UYVY and YUY2
+- A420 → RGB
+
+Less jitter when waiting on the system clock
+
+- Better system clock wait accuracy, less jitter: where available,
+ clock_nanosleep is used for higher accuracy for waits below 500
+ usecs, and waits below 2ms will first use the regular waiting system
+ and then clock_nanosleep for the remainder. The various wait
+ implementation have a latency ranging from 50 to 500+ microseconds.
+ While this is not a major issue when dealing with a low number of
+ waits per second (for ex: video), it does introduce a non-negligible
+ jitter for synchronisation of higher packet rate systems.
+
+Video decoder subframe support
-- this section will be filled in in due course
+- The GstVideoDecoder base class gained API to process input at the
+ sub-frame level. That way video decoders can start decoding slices
+ before they have received the full input frame in its entirety (to
+ the extent this is supported by the codec, of course). This helps
+ with CPU utilisation and reduces latency.
-Tracing framework and debugging improvements
+- This functionality is now being used in the OpenJPEG JPEG 2000
+ decoder, the FFmpeg H.264 decoder (in case of NAL-aligned input) and
+ the OpenMAX H.264/H.265 decoders (in case of NAL-aligned input).
-- this section will be filled in in due course
+Miscellaneous other changes and enhancements
+
+- GstDeviceMonitor no longer fails to start just because one of the
+ device providers failed to start. That could happen for example on
+ systems where the pulseaudio device provider is installed, but
+ pulseaudio isn’t actually running but ALSA is used for audio
+ instead. In the same vein the device monitor now keeps track of
+ which providers have been started (via the new
+ gst_device_provider_is_started()) and only stops actually running
+ device providers when stopping the device monitor.
+
+- On embedded systems it can be useful to create a registry that can
+ be shared and read by multiple processes running as different users.
+ It is now possible to set the new GST_REGISTRY_MODE environment
+ variable to specify the file mode for the registry file, which by
+ default is set to be only user readable/writable.
+
+- GstNetClientClock will signal lost sync in case the remote time
+ resets (e.g. because device power cycles), by emitting the “synced”
+ signal with synced=FALSE parameter, so applications can take action.
+
+- gst_value_deserialize_with_pspec() allows deserialization with a
+ hint for what the target GType should be. This allows for example
+ passing arrays of flags through the command line or
+ gst_util_set_object_arg(), eg: foo="<bar,bar+baz>".
+
+- It’s now allowed to create an empty GstVideoOverlayComposition
+ without any rectangles by passing a NULL rectangle to
+ gst_video_overlay_composition_new(). This is useful for bindings and
+ simplifies application code in some places.
+
+Tracing framework, debugging and testing improvements
+
+- New factories tracer to list loaded elements (and other plugin
+ features). This can be useful to collect a list of elements needed
+ for an application, which then in turn can be used to create a
+ tailored minimal GStreamer build that contains just the elements
+ needed and nothing else.
+- New plugin-feature-loaded tracing hook for use by tracers like the
+ new factories tracer
+
+- GstHarness: Add gst_harness_set_live() so that harnesses can be set
+ to non-live and return is-live=false in latency queries if needed.
+ Default behaviour is to always return is-live=true in latency
+ queries.
+
+- navseek: new "hold-eos" property. When enabled, the element will
+ hold back an EOS event until the next keystroke (via navigation
+ events). This can be used to keep a video sink showing the last
+ frame of a video pipeline until a key is pressed instead of tearing
+ it down immediately on EOS.
+
+- New fakeaudiosink element: mimics an audio sink and can be used for
+ testing and CI pipelines on systems where no audio system is
+ installed or running. It differs from fakesink in that it only
+ support audio caps and syncs to the clock by default like a normal
+ audio sink. It also implements the GstStreamVolume interface like
+ most audio sinks do.
+
+- New videocodectestsink element for video codec conformance testing:
+ Calculates MD5 checksums for video frames and skips any padding
+ whilst doing so. Can optionally also write back the video data with
+ padding removed into a file for easy byte-by-byte comparison with
+ reference data.
Tools
-- this section will be filled in in due course
+gst-inspect-1.0
+
+- Can sort the list of plugins by passing --sort=name as command line
+ option
+
+gst-launch-1.0
+
+- will now error out on top-level properties that don’t exist and
+ which were silently ignored before
+- On Windows the high-resolution clock is enabled now, which provides
+ better clock and timer performance on Windows (see Windows section
+ below for more details).
+
+gst-play-1.0
+
+- New --start-position command line argument to start playback from
+ the specified position
+- Audio can be muted/unmuted in interactive mode by pressing the m
+ key.
+- On Windows the high-resolution clock is enabled now (see Windows
+ section below for more details)
+
+gst-device-monitor-1.0
+
+- New --include-hidden command line argument to also show “hidden”
+ device providers
+
+ges-launch-1.0
+
+- New interactive mode that allows seeking and such. Can be disabled
+ by passing the --no-interactive argument on the command line.
+- Option to forward tags
+- Allow using an existing clip to determine the rendering format (both
+ topology and profile) via new --profile-from command line argument.
GStreamer RTSP server
-- this section will be filled in in due course
+- GstRTSPMediaFactory gained API to disable RTCP
+ (gst_rtsp_media_factory_set_enable_rtcp(), "enable-rtcp" property).
+ Previously RTCP was always allowed for all RTSP medias. With this
+ change it is possible to disable RTCP completely, no matter if the
+ client wants to do RTCP or not.
+
+- Make a mount point of / work correctly. While not allowed by the
+ RTSP 2 spec, the RTSP 1 spec is silent on this and it is used in the
+ wild. It is now possible to use / as a mount path in
+ gst-rtsp-server, e.g. rtsp://example.com/ would work with this now.
+ Note that query/fragment parts of the URI are not necessarily
+ correctly handled, and behaviour will differ between various
+ client/server implementations; so use it if you must but don’t bug
+ us if it doesn’t work with third party clients as you’d hoped.
+
+- multithreading fixes (races, refcounting issues, deadlocks)
+
+- ONVIF audio backchannel fixes
+
+- ONVIF trick mode optimisations
+
+- rtspclientsink: new "update-sdp" signal that allows updating the SDP
+ before sending it to the server via ANNOUNCE. This can be used to
+ add additional metadata to the SDP, for example. The order and
+ number of medias must not be changed, however.
GStreamer VAAPI
-- this section will be filled in in due course
+- new AV1 decoder element (vaapiav1dec)
+
+- H264 decoder: handle stereoscopic 3D video with frame packing
+ arrangement SEI messages
+
+- H265 encoder: added Screen Content Coding extensions support
+
+- H265 decoder: gained MAIN_444_12 profile support (decoded to
+ Y412_LE), and 4:2:2 12-bits support (decoded to Y212_LE)
+
+- vaapipostproc: gained BT2020 color standard support
+
+- vaapidecode: now generates caps templates dynamically at runtime in
+ order to advertise actually supported caps instead of all
+ theoretically supported caps.
+
+- GST_VAAPI_DRM_DEVICE environment variable to force a specified DRM
+ device when a DRM display is used. It is ignored when other types of
+ displays are used. By default /dev/dri/renderD128 is used for DRM
+ display.
GStreamer OMX
-- this section will be filled in in due course
+- subframe support in H.264/H.265 decoders
GStreamer Editing Services and NLE
-- this section will be filled in in due course
+- framepositioner: new "operator" property to access blending modes in
+ the compositor
+- timeline: Implement snapping to markers
+- smart-mixer: Add support for d3d11compositor and glvideomixer
+- titleclip: add "draw-shadow" child property
+- ges:// URI support to define a timeline from a description.
+- command-line-formatter
+ - Add track management to timeline description
+ - Add keyframe support
+- ges-launch-1.0:
+ - Add an interactive mode where we can seek etc…
+ - Add option to forward tags
+ - Allow using an existing clip to determine the rendering format
+ (both topology and profile) via new --profile-from command line
+ argument.
+- Fix static build
GStreamer validate
-- this section will be filled in in due course
+- report: Add a way to force backtraces on reports even if not a
+ critical issue (GST_VALIDATE_ISSUE_FLAGS_FORCE_BACKTRACE)
+- Add a flag to gst_validate_replace_variables_in_string() allow
+ defining how to resolve variables in structs
+- Add gst_validate_bin_monitor_get_scenario() to get the bin monitor
+ scenario, which is useful for applications that use Validate
+ directly.
+- Add an expected-values parameter to wait, message-type=XX allowing
+ more precise filtering of the message we are waiting for.
+- Add config file support: each test can now use a config file for the
+ given media file used to test.
+- Add support to check properties of object properties
+- scenario: Add an "action-done" signal to signal when an action is
+ done
+- scenario: Add a "run-command" action type
+- scenario: Allow forcing running action on idle from scenario file
+- scenario: Allow iterating over arrays in foreach
+- scenario: Rename ‘interlaced’ action to ‘non-blocking’
+- scenario: Add a non-blocking flag to the wait signal
GStreamer Python Bindings
-- this section will be filled in in due course
+- Fixes for Python 3.10
+- Various build fixes
+- at least one known breaking change caused by g-i annotation changes
+ (see below)
GStreamer C# Bindings
-- this section will be filled in in due course
+- Fix GstDebugGraphDetails enum
+- Updated to latests GtkSharp
+- Updated to include GStreamer 1.20 API
GStreamer Rust Bindings and Rust Plugins
-The GStreamer Rust bindings are released separately with a different
-release cadence that’s tied to gtk-rs, but the latest release has
-already been updated for the upcoming new GStreamer 1.20 API.
-
-gst-plugins-rs, the module containing GStreamer plugins written in Rust,
-has also seen lots of activity with many new elements and plugins.
-
-What follows is a list of elements and plugins available in
-gst-plugins-rs, so people don’t miss out on all those potentially useful
-elements that have no C equivalent.
-
-- FIXME: add new elements
-
-Rust audio plugins
-
-- audiornnoise: New element for audio denoising which implements the
- noise removal algorithm of the Xiph RNNoise library, in Rust
-- rsaudioecho: Port of the audioecho element from gst-plugins-good
- rsaudioloudnorm: Live audio loudness normalization element based on
- the FFmpeg af_loudnorm filter
-- claxondec: FLAC lossless audio codec decoder element based on the
- pure-Rust claxon implementation
-- csoundfilter: Audio filter that can use any filter defined via the
- Csound audio programming language
-- lewtondec: Vorbis audio decoder element based on the pure-Rust
- lewton implementation
-
-Rust video plugins
-
-- cdgdec/cdgparse: Decoder and parser for the CD+G video codec based
- on a pure-Rust CD+G implementation, used for example by karaoke CDs
-- cea608overlay: CEA-608 Closed Captions overlay element
-- cea608tott: CEA-608 Closed Captions to timed-text (e.g. VTT or SRT
- subtitles) converter
-- tttocea608: CEA-608 Closed Captions from timed-text converter
-- mccenc/mccparse: MacCaption Closed Caption format encoder and parser
-- sccenc/sccparse: Scenarist Closed Caption format encoder and parser
-- dav1dec: AV1 video decoder based on the dav1d decoder implementation
- by the VLC project
-- rav1enc: AV1 video encoder based on the fast and pure-Rust rav1e
- encoder implementation
-- rsflvdemux: Alternative to the flvdemux FLV demuxer element from
- gst-plugins-good, not feature-equivalent yet
-- rsgifenc/rspngenc: GIF/PNG encoder elements based on the pure-Rust
- implementations by the image-rs project
-
-Rust text plugins
-
-- textwrap: Element for line-wrapping timed text (e.g. subtitles) for
- better screen-fitting, including hyphenation support for some
- languages
-
-Rust network plugins
-
-- reqwesthttpsrc: HTTP(S) source element based on the Rust
- reqwest/hyper HTTP implementations and almost feature-equivalent
- with the main GStreamer HTTP source souphttpsrc
-- s3src/s3sink: Source/sink element for the Amazon S3 cloud storage
-- awstranscriber: Live audio to timed text transcription element using
- the Amazon AWS Transcribe API
-
-Generic Rust plugins
-
-- sodiumencrypter/sodiumdecrypter: Encryption/decryption element based
- on libsodium/NaCl
-- togglerecord: Recording element that allows to pause/resume
- recordings easily and considers keyframe boundaries
-- fallbackswitch/fallbacksrc: Elements for handling potentially
- failing (network) sources, restarting them on errors/timeout and
- showing a fallback stream instead
-- threadshare: Set of elements that provide alternatives for various
- existing GStreamer elements but allow to share the streaming threads
- between each other to reduce the number of threads
-- rsfilesrc/rsfilesink: File source/sink elements as replacements for
- the existing filesrc/filesink elements
+- The GStreamer Rust bindings are released separately with a different
+ release cadence that’s tied to gtk-rs, but the latest release has
+ already been updated for the upcoming new GStreamer 1.20 API (v1_20
+ feature).
+
+- gst-plugins-rs, the module containing GStreamer plugins written in
+ Rust, has also seen lots of activity with many new elements and
+ plugins. See the New Elements section above for a list of new Rust
+ elements.
Build and Dependencies
- Meson 0.59 or newer is required to build GStreamer now.
-- FIXME: this section will be filled in in due course
+- The GLib requirement has been bumped to GLib 2.56 or newer (from
+ March 2018).
+
+- The wpe plugin now requires wpe >= 2.28 and wpebackend-fdo >= 1.8
Explicit opt-in required for build of certain plugins with (A)GPL dependencies
See Building plugins with (A)GPL-licensed dependencies for more details
and a non-exhaustive list of plugins affected.
-gst-build: replaced by Monorepo
+gst-build: replaced by mono repository
-- this section will be filled in in due course
-
-- FIXME: describe + link to Monorepo FAQ
+See mono repository section above and the GStreamer mono repository FAQ.
Cerbero
on platforms where dependencies are not readily available, such as
Windows, Android, iOS and macOS.
-General improvements
+General Cerbero improvements
-- this section will be filled in in due course
+- Plugin removed: libvisual
+- New plugins: rtpmanagerbad and rist
-macOS / iOS
+macOS / iOS specific Cerbero improvements
-- this section will be filled in in due course
+- XCode 12 support
+- macOS OS release support is now future-proof, similar to iOS
+- macOS Apple Silicon (ARM64) cross-compile support has been added
+- macOS Apple Silicon (ARM64) native support is currently experimental
-Windows
+Windows specific Cerbero improvements
-- this section will be filled in in due course
+- Visual Studio 2022 support has been added
+- bootstrap is faster since it requires building fewer build-tools
+ recipes on Windows
+- package is faster due to better scheduling of recipe stages and
+ elimination of unnecessary autotools regeneration
+- The following plugins are no longer built on Windows:
+ - a52dec (another decoder is still available in libav)
+ - dvdread
+ - resindvd
Windows MSI installer
-- this section will be filled in in due course
+- no major changes
-Linux
+Linux specific Cerbero improvements
-- this section will be filled in in due course
+- Fedora, Debian OS release support is now more future-proof
+- Amazon Linux 2 support has been added
-Android
+Android specific Cerbero improvements
-- this section will be filled in in due course
+- no major changes
Platform-specific changes and improvements
Android
-- this section will be filled in in due course
+- No major changes
macOS and iOS
-- this section will be filled in in due course
+- applemedia: add ProRes support to vtenc and vtdec
Windows
-- this section will be filled in in due course
+- On Windows the high-resolution clock is enabled now in the
+ gst-launch-1.0 and gst-play-1.0 command line tools, which provides
+ better clock and timer performance on Windows, at the cost of higher
+ power consumption. By default, without the high-resolution clock
+ enabled, the timer precision on Windows is system-dependent and may
+ be as bad as 15ms which is not good enough for many multimedia
+ applications. Developers may want to do the same in their Windows
+ applications if they think it’s a good idea for their application
+ use case, and depending on the Windows version they target. This is
+ not done automatically by GStreamer because on older Windows
+ versions (pre-Windows 10) this affects a global Windows setting and
+ also there’s a power consumption vs. performance trade-off that may
+ differ from application to application.
+
+- dxgiscreencapsrc now supports resolution changes
+
+- The wasapi2 audio plugin was rewritten and now has a higher rank
+ than the old wasapi plugin since it has a number of additional
+ features such as automatic stream routing, and no
+ known-but-hard-to-fix issues. The plugin is always built if the
+ Windows 10 SDK is available now.
+
+- The wasapi device providers now detect and notify dynamic device
+ additions/removals
+
+- d3d11screencapturesrc: new desktop capture element, including
+ GstDeviceProvider implementation to enumerate/select target monitors
+ for capture.
+
+- Direct3D11/DXVA decoder now supports AV1 and MPEG2 codecs
+ (d3d11av1dec, d3d11mpeg2dec)
+
+- VP9 decoding got more reliable and stable thanks to a newly written
+ codec parser
+
+- Support for decoding interlaced H.264/AVC streams
+
+- Hardware-accelerated video deinterlacing (d3d11deinterlace) and
+ video mixing (d3d11compositor)
+
+- Video mixing with the Direct3D11 API (d3d11compositor)
+
+- MediaFoundation API based hardware encoders gained the ability to
+ receive Direct3D11 textures as an input
+
+- Seungha’s blog post “GStreamer ❤ Windows: A primer on the cool stuff
+ you’ll find in the 1.20 release” describes many of the
+ Windows-related improvements in more detail
Linux
-- this section will be filled in in due course
+- bluez: LDAC Bluetooth audio codec support in a2dpsink and avdtpsink,
+ as well as an LDAC RTP payloader (rtpldacpay) and an LDAC audio
+ encoder (ldacenc)
+
+- kmssink: gained support for NV24, NV61, RGB16/BGR16 formats;
+ auto-detect NVIDIA Tegra driver
Documentation improvements
-- this section will be filled in in due course
+- hardware-accelerated GPU plugins will now no longer always list all
+ the element variants for all available GPUs, since those are
+ system-dependent and it’s confusing for users to see those in the
+ documentation just because the GStreamer developer who generated the
+ docs had multiple GPUs to play with at the time. Instead just show
+ the default elements.
-Possibly Breaking Changes
+Possibly Breaking and Other Noteworthy Behavioural Changes
+
+- gst_parse_launch(), gst_parse_bin_from_description() and friends
+ will now error out when setting properties that don’t exist on
+ top-level bins. They were silently ignored before.
+
+- The GstWebRTC library does not expose any objects anymore with
+ public fields. Instead properties have been added to replace that
+ functionality. If you are accessing such fields in your application,
+ switch to the corresponding properties.
-- FIXME: this section will be filled in in due course
-- MPEG-TS SCTE-35 API changes (FIXME: flesh out)
-- gst_parse_launch() and friends now error out on non-existing
- properties on top-level bins where they would silently fail and
- ignore those before.
- playbin and uridecodebin now emit the source-setup signal before the
element is added to the bin and linked so that the source element is
already configured before any scheduling query comes in, which is
- useful for elements such as appsrc or giostreamsrc. (Merge Request)
+ useful for elements such as appsrc or giostreamsrc.
+
+- The source element inside urisourcebin (used inside uridecodebin3
+ which is used inside playbin3) is no longer called "source". This
+ shouldn’t affect anyone hopefully, because there’s a "setup-source"
+ signal to configure the source element and no one should rely on
+ names of internal elements anyway.
+
+- The vp8enc element now expects bps (bits per second) for the
+ "temporal-scalability-target-bitrate" property, which is consistent
+ with the "target-bitrate" property. Since additional configuration
+ is required with modern libvpx to make temporal scaling work anyway,
+ chances are that very few people will have been using this property
+
+- vp8enc and vp9enc now default to “good quality” for the "deadline"
+ property rather then “best quality”. Having the deadline set to best
+ quality causes the encoder to be absurdly slow, most real-life users
+ will want the good quality tradeoff instead.
+
+- The experimental GstTranscoder library API in gst-plugins-bad was
+ changed from a GObject signal-based notification mechanism to a
+ GstBus/message-based mechanism akin to GstPlayer/GstPlay.
+
+- MPEG-TS SCTE-35 API: semantic change for SCTE-35 splice commands:
+ timestamps passed by the application should be in running time now,
+ since users of the API can’t really be expected to predict the local
+ PTS of the muxer.
+
+- The GstContext used by souphttpsrc to share the session between
+ multiple element instances has changed. Previously it provided
+ direct access to the internal SoupSession object, now it only
+ provides access to an opaque, internal type. This change is
+ necessary because SoupSession is not thread-safe at all and can’t be
+ shared safely between arbitrary external code and souphttpsrc.
+
+- Python bindings: GObject-introspection related Annotation fixes have
+ led to a case of a GstVideo.VideoInfo-related function signature
+ changing in the Python bindings (possibly one or two other cases
+ too). This is for a function that should never have been exposed in
+ the first place though, so the bindings are being updated to throw
+ an exception in that case, and the correct replacement API has been
+ added in form of an override.
Known Issues
-- this section will be filled in in due course
-
-- There are a couple of known WebRTC-related regressions/blockers:
-
- - webrtc: DTLS setup with Chrome is broken
- - webrtcbin: First keyframe is usually lost
+- nothing in particular at this point (but also see possibly breaking
+ changes section above)
Contributors
-- this section will be filled in in due course
+Aaron Boxer, Adam Leppky, Adam Williamson, Alba Mendez, Alejandro
+González, Aleksandr Slobodeniuk, Alexander Vandenbulcke, Alex Ashley,
+Alicia Boya García, Andika Triwidada, Andoni Morales Alastruey, Andrew
+Wesie, Andrey Moiseev, Antonio Ospite, Antonio Rojas, Arthur Crippa
+Búrigo, Arun Raghavan, Ashley Brighthope, Axel Kellermann, Baek, Bastien
+Nocera, Bastien Reboulet, Benjamin Gaignard, Bing Song, Binh Truong,
+Biswapriyo Nath, Brad Hards, Brad Smith, Brady J. Garvin, Branko
+Subasic, Camilo Celis Guzman, Chris Bass, ChrisDuncanAnyvision, Chris
+White, Corentin Damman, Daniel Almeida, Daniel Knobe, Daniel Stone,
+david, David Fernandez, David Keijser, David Phung, Devarsh Thakkar,
+Dinesh Manajipet, Dmitry Samoylov, Dmitry Shusharin, Dominique Martinet,
+Doug Nazar, Ederson de Souza, Edward Hervey, Emmanuel Gil Peyrot,
+Enrique Ocaña González, Ezequiel Garcia, Fabian Orccon, Fabrice
+Fontaine, Fernando Jimenez Moreno, Florian Karydes, Francisco Javier
+Velázquez-García, François Laignel, Frederich Munch, Fredrik Pålsson,
+George Kiagiadakis, Georg Lippitsch, Göran Jönsson, Guido Günther,
+Guillaume Desmottes, Guiqin Zou, Haakon Sporsheim, Haelwenn (lanodan)
+Monnier, Haihao Xiang, Haihua Hu, Havard Graff, He Junyan, Helmut
+Januschka, Henry Wilkes, Hosang Lee, Hou Qi, Ignacio Casal Quinteiro,
+Igor Kovalenko, Ilya Kreymer, Imanol Fernandez, Jacek Tomaszewski, Jade
+Macho, Jakub Adam, Jakub Janků, Jan Alexander Steffens (heftig), Jan
+Schmidt, Jason Carrete, Jason Pereira, Jay Douglass, Jeongki Kim, Jérôme
+Laheurte, Jimmi Holst Christensen, Johan Sternerup, John Hassell, John
+Lindgren, John-Mark Bell, Jonathan Matthew, Jordan Petridis, Jose
+Quaresma, Julian Bouzas, Julien, Kai Uwe Broulik, Kasper Steensig
+Jensen, Kellermann Axel, Kevin Song, Khem Raj, Knut Inge Hvidsten, Knut
+Saastad, Kristofer Björkström, Lars Lundqvist, Lawrence Troup, Lim Siew
+Hoon, Lucas Stach, Ludvig Rappe, Luis Paulo Fernandes de Barros, Luke
+Yelavich, Mads Buvik Sandvei, Marc Leeman, Marco Felsch, Marek Vasut,
+Marian Cichy, Marijn Suijten, Marius Vlad, Markus Ebner, Mart Raudsepp,
+Matej Knopp, Mathieu Duponchelle, Matthew Waters, Matthieu De Beule,
+Mengkejiergeli Ba, Michael de Gans, Michael Olbrich, Michael Tretter,
+Michal Dzik, Miguel Paris, Mikhail Fludkov, mkba, Nazar Mokrynskyi,
+Nicholas Jackson, Nicola Murino, Nicolas Dufresne, Niklas Hambüchen,
+Nikolay Sivov, Nirbheek Chauhan, Olivier Blin, Olivier Crete, Olivier
+Crête, Paul Goulpié, Per Förlin, Peter Boba, P H, Philippe Normand,
+Philipp Zabel, Pieter Willem Jordaan, Piotrek Brzeziński, Rafał
+Dzięgiel, Rafostar, raghavendra, Raghavendra, Raju Babannavar, Raleigh
+Littles III, Randy Li, Randy Li (ayaka), Ratchanan Srirattanamet, Raul
+Tambre, reed.lawrence, Ricky Tang, Robert Rosengren, Robert Swain, Robin
+Burchell, Roman Sivriver, R S Nikhil Krishna, Ruben Gonzalez, Ruslan
+Khamidullin, Sanchayan Maity, Scott Moreau, Sebastian Dröge, Sergei
+Kovalev, Seungha Yang, Sid Sethupathi, sohwan.park, Sonny Piers, Staz M,
+Stefan Brüns, Stéphane Cerveau, Stephan Hesse, Stian Selnes, Stirling
+Westrup, Théo MAILLART, Thibault Saunier, Tim, Timo Wischer, Tim-Philipp
+Müller, Tim Schneider, Tobias Ronge, Tom Schoonjans, Tulio Beloqui,
+tyler-aicradle, U. Artie Eoff, Ung, Val Doroshchuk, VaL Doroshchuk,
+Víctor Manuel Jáquez Leal, Vivek R, Vivia Nikolaidou, Vivienne
+Watermeier, Vladimir Menshakov, Will Miller, Wim Taymans, Xabier
+Rodriguez Calvar, Xavier Claessens, Xℹ Ruoyao, Yacine Bandou, Yinhang
+Liu, youngh.lee, youngsoo.lee, yychao, Zebediah Figura, Zhang yuankun,
+Zhang Yuankun, Zhao, Zhao Zhili, , Aleksandar Topic, Antonio Ospite,
+Bastien Nocera, Benjamin Gaignard, Brad Hards, Carlos Falgueras García,
+Célestin Marot, Corentin Damman, Corentin Noël, Daniel Almeida, Daniel
+Knobe, Danny Smith, Dave Piché, Dmitry Osipenko, Fabrice Fontaine,
+fjmax, Florian Zwoch, Guillaume Desmottes, Haihua Hu, Heinrich Kruger,
+He Junyan, Jakub Adam, James Cowgill, Jan Alexander Steffens (heftig),
+Jean Felder, Jeongki Kim, Jiri Uncovsky, Joe Todd, Jordan Petridis,
+Krystian Wojtas, Marc-André Lureau, Marcin Kolny, Marc Leeman, Mark
+Nauwelaerts, Martin Reboredo, Mathieu Duponchelle, Matthew Waters,
+Mengkejiergeli Ba, Michael Gruner, Nicolas Dufresne, Nirbheek Chauhan,
+Olivier Crête, Philippe Normand, Rafał Dzięgiel, Ralf Sippl, Robert
+Mader, Sanchayan Maity, Sangchul Lee, Sebastian Dröge, Seungha Yang,
+Stéphane Cerveau, Teh Yule Kim, Thibault Saunier, Thomas Klausner, Timo
+Wischer, Tim-Philipp Müller, Tobias Reineke, Tomasz Andrzejak, Trung Do,
+Tyler Compton, Ung, Víctor Manuel Jáquez Leal, Vivia Nikolaidou, Wim
+Taymans, wngecn, Wonchul Lee, wuchang li, Xavier Claessens, Xi Ruoyao,
+Yoshiharu Hirose, Zhao,
… and many others who have contributed bug reports, translations, sent
suggestions or helped testing.
1.20.0
-1.20.0 is scheduled to be released around October/November 2021.
+1.20.0 is scheduled to be released around early February 2022.
Schedule for 1.22
unstable development version leading up to the stable 1.22 release. The
development of 1.21/1.22 will happen in the git main branch.
-The plan for the 1.22 development cycle is yet to be confirmed.
+The plan for the 1.22 development cycle is yet to be confirmed. Assuming
+no major project-wide reorganisations in the 1.22 cycle we might try and
+aim for a release around August 2022.
1.22 will be backwards-compatible to the stable 1.20, 1.18, 1.16, 1.14,
1.12, 1.10, 1.8, 1.6, 1.4, 1.2 and 1.0 release series.
------------------------------------------------------------------------
These release notes have been prepared by Tim-Philipp Müller with
-contributions from …
+contributions from Matthew Waters, Nicolas Dufresne, Nirbheek Chauhan,
+Sebastian Dröge and Seungha Yang.
License: CC BY-SA 4.0
-This is GStreamer gst-editing-services 1.19.3.
+This is GStreamer gst-editing-services 1.19.90.
GStreamer 1.19 is the development branch leading up to the next major
stable version which will be 1.20.
<release>
<Version>
+ <revision>1.19.90</revision>
+ <branch>main</branch>
+ <name></name>
+ <created>2022-01-28</created>
+ <file-release rdf:resource="https://gstreamer.freedesktop.org/src/gst-editing-services/gst-editing-services-1.19.90.tar.xz" />
+ </Version>
+ </release>
+
+ <release>
+ <Version>
<revision>1.19.3</revision>
<branch>main</branch>
<name></name>
project('gst-editing-services', 'c',
- version : '1.19.3.1',
+ version : '1.19.90',
meson_version : '>= 0.59',
default_options : [ 'warning_level=1',
'buildtype=debugoptimized' ])
-project('gst-examples', 'c', version : '1.19.3.1', license : 'LGPL')
+project('gst-examples', 'c', version : '1.19.90', license : 'LGPL')
cc = meson.get_compiler('c')
m_dep = cc.find_library('m', required : false)
-project('gst-integration-testsuites', [], version: '1.19.3.1', meson_version : '>= 0.59', license: 'LGPL')
+project('gst-integration-testsuites', [], version: '1.19.90', meson_version : '>= 0.59', license: 'LGPL')
+=== release 1.19.90 ===
+
+2022-01-28 14:28:35 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * NEWS:
+ * RELEASE:
+ * gst-libav.doap:
+ * meson.build:
+ Release 1.19.90
+
+2022-01-28 14:28:28 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * ChangeLog:
+ Update ChangeLogs for 1.19.90
+
2022-01-17 01:33:47 +0800 Xi Ruoyao <xry111@mengyan1223.wang>
* ext/libav/gstavauddec.c:
GStreamer 1.20 Release Notes
GStreamer 1.20 has not been released yet. It is scheduled for release in
-November 2021.
+late January / early February 2022.
1.19.x is the unstable development version that is being developed in
-the git main branch and which will eventually result in 1.20, and 1.19.3
-is the current development release in that series
-
-It is expected that feature freeze will be in early October 2021,
-followed by one or two 1.19.9x pre-releases and the new 1.20 stable
-release around October/November 2021.
+the git main branch and which will eventually result in 1.20, and
+1.19.90 is the first release candidate in that series (1.20rc1).
1.20 will be backwards-compatible to the stable 1.18, 1.16, 1.14, 1.12,
-1.10, 1.8, 1.6,, 1.4, 1.2 and 1.0 release series.
+1.10, 1.8, 1.6, 1.4, 1.2 and 1.0 release series.
See https://gstreamer.freedesktop.org/releases/1.20/ for the latest
version of this document.
-Last updated: Monday 1 November 2021, 01:00 UTC (log)
+Last updated: Wednesday 26 January 2022, 01:00 UTC (log)
Introduction
Highlights
-- this section will be completed in due course
+- Development in GitLab was switched to a single git repository
+ containing all the modules
+- GstPlay: new high-level playback library, replaces GstPlayer
+- WebM Alpha decoding support
+- Encoding profiles can now be tweaked with additional
+ application-specified element properties
+- Compositor: multi-threaded video conversion and mixing
+- RTP header extensions: unified support in RTP depayloader and
+ payloader base classes
+- SMPTE 2022-1 2-D Forward Error Correction support
+- Smart encoding (passthrough) support for VP8, VP9, H.265 in
+ encodebin and transcodebin
+- Runtime compatibility support for libsoup2 and libsoup3 (libsoup3
+ support experimental)
+- Video decoder subframe support
+- Video decoder automatic packet-loss, data corruption, and keyframe
+ request handling for RTP / WebRTC / RTSP
+- MP4 and Matroska muxers now support profile/level/resolution changes
+ for H264/H265 input streams (i.e. codec data changing on the fly)
+- MP4 muxing mode that initially creates a fragmented mp4 which is
+ converted to a regular mp4 on EOS
+- Audio support for the WebKit Port for Embedded (WPE) web page source
+ element
+- CUDA based video color space convert and rescale elements and
+ upload/download elements
+- NVIDIA memory:NVMM support for OpenGL glupload and gldownload
+ elements
+- Many WebRTC improvements
+- The new VA-API plugin implemention fleshed out with more decoders
+ and new postproc elements
+- AppSink API to retrieve events in addition to buffers and buffer
+ lists
+- AppSrc gained more configuration options for the internal queue
+ (leakiness, limits in buffers and time, getters to read current
+ levels)
+- Updated Rust bindings and many new Rust plugins
+- Improved support for custom minimal GStreamer builds
+- Support build against FFmpeg 5.0
+- Linux Stateless CODEC support gained MPEG2 and VP9
+- Windows Direct3D11/DXVA decoder gained AV1 and MPEG2 support
+- Lots of new plugins, features, performance improvements and bug
+ fixes
Major new features and changes
Noteworthy new features and API
-- this section will be filled in in due course
+- gst_element_get_request_pad() has been deprecated in favour of the
+ newly-added gst_element_request_pad_simple() which does the exact
+ same thing but has a less confusing name that hopefully makes clear
+ that the function request a new pad rather than just retrieves an
+ already-existing request pad.
+
+Development in GitLab was switched to a single git repository containing all the modules
+
+The GStreamer multimedia framework is a set of libraries and plugins
+split into a number of distinct modules which are released independently
+and which have so far been developed in separate git repositories in
+freedesktop.org GitLab.
+
+In addition to these separate git repositories there was a gst-build
+module that would use the Meson build systems’s subproject feature to
+download each individual module and then build everything in one go. It
+would also provide an uninstalled development environment that made it
+easy to work on GStreamer and use or test versions other than the
+system-installed GStreamer version.
+
+All of these modules have now (as of 28 September 2021) been merged into
+a single git repository (“Mono repository” or “monorepo”) which should
+simplify development workflows and continuous integration, especially
+where changes need to be made to multiple modules at once.
+
+This mono repository merge will primarily affect GStreamer developers
+and contributors and anyone who has workflows based on the GStreamer git
+repositories.
+
+The Rust bindings and Rust plugins modules have not been merged into the
+mono repository at this time because they follow a different release
+cycle.
+
+The mono repository lives in the existing GStreamer core git repository
+in GitLab in the new main branch and all future development will happen
+on this branch.
+
+Modules will continue to be released as separate tarballs.
+
+For more details, please see the GStreamer mono repository FAQ.
+
+GstPlay: new high-level playback library replacing GstPlayer
+
+- GstPlay is a new high-level playback library that replaces the older
+ GstPlayer API. It is basically the same API as GstPlayer but
+ refactored to use bus messages for application notifications instead
+ of GObject signals. There is still a signal adapter object for those
+ who prefer signals. Since the existing GstPlayer API is already in
+ use in various applications, it didn’t seem like a good idea to
+ break it entirely. Instead a new API was added, and it is expected
+ that this new GstPlay API will be moved to gst-plugins-base in
+ future.
+
+- The existing GstPlayer API is scheduled for deprecation and will be
+ removed at some point in the future (e.g. in GStreamer 1.24), so
+ application developers are urged to migrate to the new GstPlay API
+ at their earliest convenience.
+
+WebM alpha decoding
+
+- Implement WebM alpha decoding (VP8/VP9 with alpha), which required
+ support and additions in various places. This is supported both with
+ software decoders and hardware-accelerated decoders.
+
+- VP8/VP9 don’t support alpha components natively in the codec, so the
+ way this is implemented in WebM is by encoding the alpha plane with
+ transparency data as a separate VP8/VP9 stream. Inside the WebM
+ container (a variant of Matroska) this is coded as a single video
+ track with the “normal” VP8/VP9 video data making up the main video
+ data and each frame of video having an encoded alpha frame attached
+ to it as extra data ("BlockAdditional").
+
+- matroskademux has been extended extract this per-frame alpha side
+ data and attach it in form of a GstVideoCodecAlphaMeta to the
+ regular video buffers. Note that this new meta is specific to this
+ VP8/VP9 alpha support and can’t be used to just add alpha support to
+ other codecs that don’t support it. Lastly, matroskademux also
+ advertises the fact that the streams contain alpha in the caps.
+
+- The new codecalpha plugin contains various bits of infrastructure to
+ support autoplugging and debugging:
+
+ - codecalphademux splits out the alpha stream from the metas on
+ the regular VP8/VP9 buffers
+ - alphacombine takes two decoded raw video streams (one alpha, one
+ the regular video) and combines it into a video stream with
+ alpha
+ - vp8alphadecodebin + vp9alphadecodebin are wrapper bins that use
+ the regular vp8dec and vp9dec software decoders to decode
+ regular and alpha streams and combine them again. To decodebin
+ these look like regular decoders which ju
+ - The V4L2 CODEC plugin has stateless VP8/VP9 decoders that can
+ decode both alpha and non-alpha stream with a single decoder
+ instance
+
+- A new AV12 video format was added which is basically NV12 with an
+ alpha plane, which is more convenient for many hardware-accelerated
+ decoders.
+
+- Watch Nicolas Dufresne’s LCA 2022 talk “Bringing WebM Alpha support
+ to GStreamer” for all the details and a demo.
+
+RTP Header Extensions Base Class and Automatic Header Extension Handling in RTP Payloaders and Depayloaders
+
+- RTP Header Extensions are specified in RFC 5285 and provide a way to
+ add small pieces of data to RTP packets in between the RTP header
+ and the RTP payload. This is often used for per-frame metadata,
+ extended timestamps or other application-specific extra data. There
+ are several commonly-used extensions specified in various RFCs, but
+ senders are free to put any kind of data in there, as long as sender
+ and receiver both know what that data is. Receivers that don’t know
+ about the header extensions will just skip the extra data without
+ ever looking at it. These header extensions can often be combined
+ with any kind of payload format, so may need to be supported by many
+ RTP payloader and depayloader elements.
+
+- Inserting and extracting RTP header extension data has so far been a
+ bit inconvenient in GStreamer: There are functions to add and
+ retrieve RTP header extension data from RTP packets, but nothing
+ works automatically, even for common extensions. People would have
+ to do the insertion/extraction either in custom elements
+ before/after the RTP payloader/depayloader, or inside pad probes,
+ which isn’t very nice.
+
+- This release adds various pieces of new infrastructure for generic
+ RTP header extension handling, as well as some implementations for
+ common extensions:
+
+ - GstRTPHeaderExtension is a new helper base class for reading and
+ writing RTP header extensions. Nominally this subclasses
+ GstElement, but only so these extensions are stored in the
+ registry where they can be looked up by URI or name. They don’t
+ have pads and don’t get added to the pipeline graph as an
+ element.
+
+ - "add-extension" and "clear-extension" action signals on RTP
+ payloaders and depayloaders for manual extension management
+
+ - The "request-extension" signal will be emitted if an extension
+ is encountered that requires explicit mapping by the application
+
+ - new "auto-header-extension" property on RTP payloaders and
+ depayloaders for automatic handling of known header extensions.
+ This is enabled by default. The extensions must be signalled via
+ caps / SDP.
+
+ - RTP header extension implementations:
+
+ - rtphdrextclientaudiolevel: Client-to-Mixer Audio Level
+ Indication (RFC 6464) (also see below)
+ - rtphdrextcolorspace: Color Space extension, extends RTP
+ packets with color space and high dynamic range (HDR)
+ information
+ - rtphdrexttwcc: Transport Wide Congestion Control support
+
+- gst_rtp_buffer_remove_extension_data() is a new helper function to
+ remove an RTP header extension from an RTP buffer
+
+- The existing gst_rtp_buffer_set_extension_data() now also supports
+ shrinking the extension data in size
+
+AppSink and AppSrc improvements
+
+- appsink: new API to pull events out of appsink in addition to
+ buffers and buffer lists.
+
+ There was previously no way for users to receive incoming events
+ from appsink properly serialised with the data flow, even if they
+ are serialised events. The reason for that is that the only way to
+ intercept events was via a pad probe on the appsink sink pad, but
+ there is also internal queuing inside of appsink, so it’s difficult
+ to ascertain the right order of everything in all cases.
+
+ There is now a new "new-serialized-event" signal which will be
+ emitted when there’s a new event pending (just like the existing
+ "new-sample" signal). The "emit-signals" property must be set to
+ TRUE in order to activate this (but it’s also fine to just pull from
+ the application thread without using the signals).
+
+ gst_app_sink_pull_object() and gst_app_sink_try_pull_object() can be
+ used to pull out either an event or a new sample carrying a buffer
+ or buffer list, whatever is next in the queue.
+
+ EOS events will be filtered and will not be returned. EOS handling
+ can be done the ususal way, same as with _pull_sample().
+
+- appsrc: allow configuration of internal queue limits in time and
+ buffers and add leaky mode.
+
+ There is internal queuing inside appsrc so the application thread
+ can push data into the element which will then be picked up by the
+ source element’s streaming thread and pushed into the pipeline from
+ that streaming thread. This queue is unlimited by default and until
+ now it was only possible to set a maximum size limit in bytes. When
+ that byte limit is reached, the pushing thread (application thread)
+ would be blocked until more space becomes available.
+
+ A limit in bytes is not particularly useful for many use cases, so
+ now it is possible to also configure limits in time and buffers
+ using the new "max-time" and "max-buffers" properties. Of course
+ there are also matching new read-only"current-level-buffers" and
+ "current-level-time properties" properties to query the current fill
+ level of the internal queue in time and buffers.
+
+ And as if that wasn’t enough the internal queue can also be
+ configured as leaky using the new "leaky-type" property. That way
+ when the queue is full the application thread won’t be blocked when
+ it tries to push in more data, but instead either the new buffer
+ will be dropped or the oldest data in the queue will be dropped.
+
+Better string serialization of nested GstCaps and GstStructures
+
+- New string serialisation format for structs and caps that can handle
+ nested structs and caps properly by using brackets to delimit nested
+ items (e.g. some-struct, some-field=[nested-struct, nested=true]).
+ Unlike the default format the new variant can also support more than
+ one level of nesting. For backwards-compatibility reasons the old
+ format is still output by default when serialising caps and structs
+ using the existing API. The new functions gst_caps_serialize() and
+ gst_structure_serialize() can be used to output strings in the new
+ format.
+
+Convenience API for custom GstMetas
+
+- New convenience API to register and create custom GstMetas:
+ gst_meta_register_custom() and gst_buffer_add_custom_meta(). Such
+ custom meta is backed by a GstStructure and does not require that
+ users of the API expose their GstMeta implementation as public API
+ for other components to make use of it. In addition, it provides a
+ simpler interface by ignoring the impl vs. api distinction that the
+ regular API exposes. This new API is meant to be the meta
+ counterpart to custom events and messages, and to be more convenient
+ than the lower-level API when the absolute best performance isn’t a
+ requirement. The reason it’s less performant than a “proper” meta is
+ that a proper meta is just a C struct in the end whereas this goes
+ through the GstStructure API which has a bit more overhead, which
+ for most scenarios is negligible however. This new API is useful for
+ experimentation or proprietary metas, but also has some limitations:
+ it can only be used if there’s a single producer of these metas;
+ it’s not allowed to register the same custom meta multiple times or
+ from multiple places.
+
+Additional Element Properties on Encoding Profiles
+
+- GstEncodingProfile: The new "element-properties" and
+ gst_encoding_profile_set_element_properties() API allows
+ applications to set additional element properties on encoding
+ profiles to configure muxers and encoders. So far the encoding
+ profile template was the only place where this could be specified,
+ but often what applications want to do is take a ready-made encoding
+ profile shipped by GStreamer or the application and then tweak the
+ settings on top of that, which is now possible with this API. Since
+ applications can’t always know in advance what encoder element will
+ be used in the end, it’s even possible to specify properties on a
+ per-element basis.
+
+ Encoding Profiles are used in the encodebin, transcodebin and
+ camerabin elements and APIs to configure output formats (containers
+ and elementary streams).
+
+Audio Level Indication Meta for RFC 6464
+
+- New GstAudioLevelMeta containing Audio Level Indication as per RFC
+ 6464
+
+- The level element has been updated to add GstAudioLevelMeta on
+ buffers if the "audio-level-meta" property is set to TRUE. This can
+ then in turn be picked up by RTP payloaders to signal the audio
+ level to receivers through RTP header extensions (see above).
+
+- New Client-to-Mixer Audio Level Indication (RFC6464) RTP Header
+ Extension which should be automatically created and used by RTP
+ payloaders and depayloaders if their "auto-header-extension"
+ property is enabled and if the extension is part of the RTP caps.
+
+Automatic packet loss, data corruption and keyframe request handling for video decoders
+
+- The GstVideoDecoder base class has gained various new APIs to
+ automatically handle packet loss and data corruption better by
+ default, especially in RTP, RTSP and WebRTC streaming scenarios, and
+ to give subclasses more control about how they want to handle
+ missing data:
+
+ - Video decoder subclasses can mark output frames as corrupted via
+ the new GST_VIDEO_CODEC_FRAME_FLAG_CORRUPTED flag
+
+ - A new "discard-corrupted-frames" property allows applications to
+ configure decoders so that corrupted frames are directly
+ discarded instead of being forwarded inside the pipeline. This
+ is a replacement for the "output-corrupt" property of the FFmpeg
+ decoders.
+
+ - RTP depayloaders can now signal to decoders that data is missing
+ when sending GAP events for lost packets. GAP events can be sent
+ for various reason in a GStreamer pipeline. Often they are just
+ used to let downstream elements know that there isn’t a buffer
+ available at the moment, so downstream elements can move on
+ instead of waiting for one. They are also sent by RTP
+ depayloaders in the case that packets are missing, however, and
+ so far a decoder was not able to differentiate the two cases.
+ This has been remedied now: GAP events can be decorated with
+ gst_event_set_gap_flags() and GST_GAP_FLAG_MISSING_DATA to let
+ decoders now what happened, and decoders can then use that in
+ some cases to handle missing data better.
+
+ - The GstVideoDecoder::handle_missing_data vfunc was added to
+ inform subclasses about packet loss or missing data and let them
+ handle it in their own way if they like.
+
+ - gst_video_decoder_set_needs_sync_point() lets subclasses signal
+ that they need the stream to start with a sync point. If
+ enabled, the base class will discard all non-sync point frames
+ in the beginning and after a flush and does not pass them to the
+ subclass. Furthermore, if the first frame is not a sync point,
+ the base class will try and request a sync frame from upstream
+ by sending a force-key-unit event (see next items).
+
+ - New "automatic-request-sync-points" and
+ "automatic-request-sync-point-flags" properties to automatically
+ request sync points when needed, e.g. on packet loss or if the
+ first frame is not a keyframe. Applications may want to enable
+ this on decoders operating in e.g. RTP/WebRTC/RTSP receiver
+ pipelines.
+
+ - The new "min-force-key-unit-interval" property can be used to
+ ensure there’s a minimal interval between keyframe requests to
+ upstream (and/or the sender) and we’re not flooding the sender
+ with key unit requests.
+
+ - gst_video_decoder_request_sync_point() allows subclasses to
+ request a new sync point (e.g. if they choose to do their own
+ missing data handling). This will still honour the
+ "min-force-key-unit-interval" property if set.
+
+Improved support for custom minimal GStreamer builds
+
+- Element registration and registration of other plugin features
+ inside plugin init functions has been improved in order to
+ facilitate minimal custom GStreamer builds.
+
+- A number of new macros have been added to declare and create
+ per-element and per-pluginfeature register functions in all plugins,
+ and then call those from the per-plugin plugin_init functions:
+
+ - GST_ELEMENT_REGISTER_DEFINE,
+ GST_DEVICE_PROVIDER_REGISTER_DEFINE,
+ GST_DYNAMIC_TYPE_REGISTER_DEFINE, GST_TYPE_FIND_REGISTER_DEFINE
+ for the actual registration call with GStreamer
+ - GST_ELEMENT_REGISTER, GST_DEVICE_PROVIDER_REGISTER,
+ GST_DYNAMIC_TYPE_REGISTER, GST_PLUGIN_STATIC_REGISTER,
+ GST_TYPE_FIND_REGISTER to call the registration function defined
+ by the REGISTER_DEFINE macro
+ - GST_ELEMENT_REGISTER_DECLARE,
+ GST_DEVICE_PROVIDER_REGISTER_DECLARE,
+ GST_DYNAMIC_TYPE_REGISTER_DECLARE,
+ GST_TYPE_FIND_REGISTER_DECLARE to declare the registration
+ function defined by the REGISTER_DEFINE macro
+ - and various variants for advanced use cases.
+
+- This means that applications can call the per-element and
+ per-pluginfeature registration functions for only the elements they
+ need instead of registering plugins as a whole with all kinds of
+ elements that may not be required (e.g. encoder and decoder instead
+ of just decoder). In case of static linking all unused functions and
+ their dependencies would be removed in this case by the linker,
+ which helps minimise binary size for custom builds.
+
+- gst_init() will automatically call a gst_init_static_plugins()
+ function if one exists.
+
+- See the GStreamer static build documentation and Stéphane’s blog
+ post Generate a minimal GStreamer build, tailored to your needs for
+ more details.
New elements
-- this section will be filled in in due course
+- New aesdec and aesenc elements for AES encryption and decryption in
+ a custom format.
+
+- New encodebin2 element with dynamic/sometimes source pads in order
+ to support the option of doing the muxing outside of encodebin,
+ e.g. in combination with a splitmuxsink.
+
+- New fakeaudiosink and videocodectestsink elements for testing and
+ debugging (see below for more details)
+
+- rtpisacpay, rtpisacdepay: new RTP payloader and depayloader for iSAC
+ audio codec
+
+- rtpst2022-1-fecdec, rtpst2022-1-fecenc: new elements providing SMPTE
+ 2022-1 2-D Forward Error Correction. More details in Mathieu’s blog
+ post.
+
+- isac: new plugin wrapping the Internet Speech Audio Codec reference
+ encoder and decoder from the WebRTC project.
+
+- asio: plugin for Steinberg ASIO (Audio Streaming Input/Output) API
+
+- gssrc, gssink: add source and sink for Google Cloud Storage
+
+- onnx: new plugin to apply ONNX neural network models to video
+
+- openaptx: aptX and aptX-HD codecs using libopenaptx (v0.2.0)
+
+- qroverlay, debugqroverlay: new elements that allows overlaying data
+ on top of video in form of a QR code
+
+- cvtracker: new OpenCV-based tracker element
+
+- av1parse, vp9parse: new parsers for AV1 and VP9 video
+
+- va: work on the new VA-API plugin implementation for
+ hardware-accelerated video decoding and encoding has continued at
+ pace, with various new decoders and filters having joined the
+ initial vah264dec:
+
+ - vah265dec: VA-API H.265 decoder
+ - vavp8dec: VA-API VP8 decoder
+ - vavp9dec: VA-API VP9 decoder
+ - vaav1dec: VA-API AV1 decoder
+ - vampeg2dec: VA-API MPEG-2 decoder
+ - vadeinterlace: : VA-API deinterlace filter
+ - vapostproc: : VA-API postproc filter (color conversion,
+ resizing, cropping, color balance, video rotation, skin tone
+ enhancement, denoise, sharpen)
+
+ See Víctor’s blog post “GstVA in GStreamer 1.20” for more details
+ and what’s coming up next.
+
+- vaapiav1dec: new AV1 decoder element (in gstreamer-vaapi)
+
+- msdkav1dec: hardware-accelerated AV1 decoder using the Intel Media
+ SDK / oneVPL
+
+- nvcodec plugin for NVIDIA NVCODEC API for hardware-accelerated video
+ encoding and decoding:
+
+ - cudaconvert, cudascale: new CUDA based video color space convert
+ and rescale elements
+ - cudaupload, cudadownload: new helper elements for memory
+ transfer between CUDA and system memory spaces
+ - nvvp8sldec, nvvp9sldec: new GstCodecs-based VP8/VP9 decoders
+
+- Various new hardware-accelerated elements for Windows:
+
+ - d3d11screencapturesrc: new desktop capture element, including a
+ GstDeviceProvider implementation to enumerate/select target
+ monitors for capture.
+ - d3d11av1dec and d3d11mpeg2dec: AV1 and MPEG-2 decoders
+ - d3d11deinterlace: deinterlacing filter
+ - d3d11compositor: video composing element
+ - see Windows section below for more details
+
+- new Rust plugins:
+
+ - audiornnoise: Removes noise from an audio stream
+ - awstranscribeparse: Parses AWS audio transcripts into timed text
+ buffers
+ - ccdetect: Detects if valid closed captions are present in a
+ closed captions stream
+ - cea608tojson: Converts CEA-608 Closed Captions to a JSON
+ representation
+ - cmafmux: CMAF fragmented MP4 muxer
+ - dashmp4mux: DASH fragmented MP4 muxer
+ - isofmp4mux: ISO fragmented MP4 muxer
+ - ebur128level: EBU R128 Loudness Level Measurement
+ - ffv1dec: FFV1 video decoder
+ - gtk4paintablesink: GTK4 video sink, which provides a
+ GdkPaintable that can be rendered in various widgets
+ - hlssink3: HTTP Live Streaming sink
+ - hrtfrender: Head-Related Transfer Function (HRTF) renderer
+ - hsvdetector: HSV colorspace detector
+ - hsvfilter: HSV colorspace filter
+ - jsongstenc: Wraps buffers containing any valid top-level JSON
+ structures into higher level JSON objects, and outputs those as
+ ndjson
+ - jsongstparse: Parses ndjson as output by jsongstenc
+ - jsontovtt: converts JSON to WebVTT subtitles
+ - regex: Applies regular expression operations on text
+ - roundedcorners: Adds rounded corners to video
+ - spotifyaudiosrc: Spotify source
+ - textahead: Display upcoming text buffers ahead (e.g. for
+ Karaoke)
+ - transcriberbin: passthrough bin that transcribes raw audio to
+ closed captions using awstranscriber and puts the captions as
+ metas onto the video
+ - tttojson: Converts timed text to a JSON representation
+ - uriplaylistbin: Playlist source bin
+ - webpdec-rs: WebP image decoder with animation support
+
+- New plugin codecalpha with elements to assist with WebM Alpha
+ decoding
+
+ - codecalphademux: Split stream with GstVideoCodecAlphaMeta into
+ two streams
+ - alphacombine: Combine two raw video stream (I420 or NV12) as one
+ stream with alpha channel (A420 or AV12)
+ - vp8alphadecodebin: A bin to handle software decoding of VP8 with
+ alpha
+ - vp9alphadecodebin: A bin to handle software decoding of VP9 with
+ alpha
+
+- New hardware accelerated elements for Linux:
+
+ - v4l2slmpeg2dec: Support for Linux Stateless MPEG2 decoders
+ - v4l2slvp9dec: Support for Linux Stateless VP9 decoders
+ - v4l2slvp8alphadecodebin: Support HW accelerated VP8 with alpha
+ layer decoding
+ - v4l2slvp9alphadecodebin: Support HW accelerated VP9 with alpha
+ layer decoding
New element features and additions
-- this section will be filled in in due course
+- assrender: handle more font mime types; better interaction with
+ matroskademux for embedded fonts
+
+- audiobuffersplit: Add support for specifying output buffer size in
+ bytes (not just duration)
+
+- audiolatency: new "samplesperbuffer" property so users can configure
+ the number of samples per buffer. The default value is 240 samples
+ which is equivalent to 5ms latency with a sample rate of 48000,
+ which might be larger than actual buffer size of audio capture
+ device.
+
+- audiomixer, audiointerleave, GstAudioAggregator: now keep a count of
+ samples that are dropped or processed as statistic and can be made
+ to post QoS messages on the bus whenever samples are dropped by
+ setting the "qos-messages" property on input pads.
+
+- audiomixer, compositor: improved handling of new inputs added at
+ runtime. New API was added to the GstAggregator base class to allow
+ subclasses to opt into an aggregation mode where inactive pads are
+ ignored when processing input buffers
+ (gst_aggregator_set_ignore_inactive_pads(),
+ gst_aggregator_pad_is_inactive()). An “inactive pad” in this context
+ is a pad which, in live mode, hasn’t yet received a first buffer,
+ but has been waited on at least once. What would happen usually in
+ this case is that the aggregator would wait for data on this pad
+ every time, up to the maximum configured latency. This would
+ inadvertently push mixer elements in live mode to the configured
+ latency envelope and delay processing when new inputs are added at
+ runtime until these inputs have actually produced data. This is
+ usually undesirable. With this new API, new inputs can be added
+ (requested) and configured and they won’t delay the data processing.
+ Applications can opt into this new behaviour by setting the
+ "ignore-inactive-pads" property on compositor, audiomixer or other
+ GstAudioAggregator-based elements.
+
+- cccombiner: implement “scheduling” of captions. So far cccombiner’s
+ behaviour was essentially that of a funnel: it strictly looked at
+ input timestamps to associate together video and caption buffers.
+ Now it will try to smoothly schedule caption buffers in order to
+ have exactly one per output video buffer. This might involve
+ rewriting input captions, for example when the input is CDP then
+ sequence counters are rewritten, time codes are dropped and
+ potentially re-injected if the input video frame had a time code
+ meta. This can also lead to the input drifting from synchronization,
+ when there isn’t enough padding in the input stream to catch up. In
+ that case the element will start dropping old caption buffers once
+ the number of buffers in its internal queue reaches a certain limit
+ (configurable via the "max-scheduled" property). The new original
+ funnel-like behaviour can be restored by setting the "scheduling"
+ property to FALSE.
+
+- ccconverter: new "cdp-mode" property to specify which sections to
+ include in CDP packets (timecode, CC data, service info). Various
+ software, including ffmpeg’s Decklink support, fails parsing CDP
+ packets that contain anything but CC data in the CDP packets.
+
+- clocksync: new "sync-to-first" property for automatic timestamp
+ offset setup: if set clocksync will set up the "ts-offset" value
+ based on the first buffer and the pipeline’s running time when the
+ first buffer arrived. The newly configured "ts-offset" in this case
+ would be the value that allows outputting the first buffer without
+ waiting on the clock. This is useful for example to feed a non-live
+ input into an already-running pipeline.
+
+- compositor:
+
+ - multi-threaded input conversion and compositing. Set the
+ "max-threads" property to activate this.
+ - new "sizing-policy" property to support display aspect ratio
+ (DAR)-aware scaling. By default the image is scaled to fill the
+ configured destination rectangle without padding and without
+ keeping the aspect ratio. With sizing-policy=keep-aspect-ratio
+ the input image is scaled to fit the destination rectangle
+ specified by GstCompositorPad:{xpos, ypos, width, height}
+ properties preserving the aspect ratio. As a result, the image
+ will be centered in the destination rectangle with padding if
+ necessary.
+ - new "zero-size-is-unscaled" property on input pads. By default
+ pad width=0 or pad height=0 mean that the stream should not be
+ scaled in that dimension. But if the "zero-size-is-unscaled"
+ property is set to FALSE a width or height of 0 is instead
+ interpreted to mean that the input image on that pad should not
+ be composited, which is useful when creating animations where an
+ input image is made smaller and smaller until it disappears.
+ - improved handling of new inputs at runtime via
+ "ignore-inactive-pads"property (see above for details)
+ - allow output format with alpha even if none of the inputs have
+ alpha (also glvideomixer and other GstVideoAggregator
+ subclasses)
+
+- dashsink: add h265 codec support and signals for allowing custom
+ playlist/fragment output
+
+- decodebin3:
+
+ - improved decoder selection, especially for hardware decoders
+ - make input activation “atomic” when adding inputs dynamically
+ - better interleave handling: take into account decoder latency
+ for interleave size
+
+- decklink:
+
+ - Updated DeckLink SDK to 11.2 to support DeckLink 8K Pro
+ - decklinkvideosrc:
+ - More accurate and stable capture timestamps: use the
+ hardware reference clock time when the frame was finished
+ being captured instead of a clock time much further down the
+ road.
+ - Automatically detect widescreen vs. normal NTSC/PAL
+
+- encodebin:
+
+ - add “smart encoding” support for H.265, VP8 and VP9 (i.e. only
+ re-encode where needed and otherwise pass through encoded video
+ as-is).
+ - H264/H265 smart encoding improvements: respect user-specified
+ stream-format, but if not specified default to avc3/hvc1 with
+ in-band SPS/PPS/VPS signalling for more flexibility.
+ - new encodebin2 element with dynamic/sometimes source pads in
+ order to support the option of doing the muxing outside of
+ encodebin, e.g. in combination with splitmuxsink.
+ - add APIs to set element properties on encoding profiles (see
+ below)
+
+- errorignore: new "ignore-eos" property to also ignore FLOW_EOS from
+ downstream elements
+
+- giosrc: add support for growing source files: applications can
+ specify that the underlying file being read is growing by setting
+ the "is-growing" property. If set, the source won’t EOS when it
+ reaches the end of the file, but will instead start monitoring it
+ and will start reading data again whenever a change is detected. The
+ new "waiting-data" and "done-waiting-data" signals keep the
+ application informed about the current state.
+
+- gtksink, gtkglsink:
+
+ - scroll event support: forwarded as navigation events into the
+ pipeline
+ - "video-aspect-ratio-override" property to force a specific
+ aspect ratio
+ - "rotate-method" property and support automatic rotation based on
+ image tags
+
+- identity: new "stats" property allows applications to retrieve the
+ number of bytes and buffers that have passed through so far.
+
+- interlace: add support for more formats, esp 10-bit, 12-bit and
+ 16-bit ones
+
+- jack: new "low-latency" property for automatic latency-optimized
+ setting and "port-names" property to select ports explicitly
+
+- jpegdec: support output conversion to RGB using libjpeg-turbo (for
+ certain input files)
+
+- line21dec:
+
+ - "mode" property to control whether and how detected closed
+ captions should be inserted in the list of existing close
+ caption metas on the input frame (if any): add, drop, or
+ replace.
+ - "ntsc-only" property to only look for captions if video has NTSC
+ resolution
+
+- line21enc: new "remove-caption-meta" to remove metas from output
+ buffers after encoding the captions into the video data; support for
+ CDP closed captions
+
+- matroskademux, matroskamux: Add support for ffv1, a lossless
+ intra-frame video coding format.
+
+- matroskamux: accept in-band SPS/PPS/VPS for H264 and H265
+ (i.e. stream-format avc3 and hev1) which allows on-the-fly
+ profile/level/resolution changes.
+
+- matroskamux: new "cluster-timestamp-offset" property, useful for use
+ cases where the container timestamps should map to some absolute
+ wall clock time, for example.
+
+- rtpsrc: add "caps" property to allow explicit setting of the caps
+ where needed
+
+- mpegts: support SCTE-35 passthrough via new "send-scte35-events"
+ property on MPEG-TS demuxer tsdemux. When enabled, SCTE 35 sections
+ (eg ad placement opportunities) are forwarded as events donwstream
+ where they can be picked up again by mpegtsmux. This required a
+ semantic change in the SCTE-35 section API: timestamps are now in
+ running time instead of muxer pts.
+
+- tsdemux: Handle PCR-less MPEG-TS streams; more robust timestamp
+ handling in certain corner cases and for poorly muxed streams.
+
+- mpegtsmux:
+
+ - More conformance improvements to make MPEG-TS analyzers happy:
+ - PCR timing accuracy: Improvements to the way mpegtsmux
+ outputs PCR observations in CBR mode, so that a PCR
+ observation is always inserted when needed, so that we never
+ miss the configured pcr-interval, as that triggers various
+ MPEG-TS analyser errors.
+ - Improved PCR/SI scheduling
+ - Don’t write PCR until PAT/PMT are output to make sure streams
+ start cleanly with a PAT/PMT.
+ - Allow overriding the automatic PMT PID selection via
+ application-supplied PMT_%d fields in the prog-map
+ structure/property.
+
+- mp4mux:
+
+ - new "first-moov-then-finalise" mode for fragmented output where
+ the output will start with a self-contained moov atom for the
+ first fragment, and then produce regular fragments. Then at the
+ end when the file is finalised, the initial moov is invalidated
+ and a new moov is written covering the entire file. This way the
+ file is a “fragmented mp4” file while it is still being written
+ out, and remains playable at all times, but at the end it is
+ turned into a regular mp4 file (with former fragment headers
+ remaining as unused junk data in the file).
+ - support H.264 avc3 and H.265 hvc1 stream formats as input where
+ the codec data is signalled in-band inside the bitstream instead
+ of caps/file headers.
+ - support profile/level/resolution changes for H264/H265 input
+ streams (i.e. codec data changing on the fly). Each codec_data
+ is put into its own SampleTableEntry inside the stsd, unless the
+ input is in avc3 stream format in which case it’s written
+ in-band an not in the headers.
+
+- multifilesink: new ""min-keyframe-distance"" property to make
+ minimum distance between keyframes in next-file=key-frame mode
+ configurable instead of hard-coding it to 10 seconds.
+
+- mxfdemux has seen a big refactoring to support non-frame wrappings
+ and more accurate timestamp/seek handling for some formats
+
+- msdk plugin for hardware-accelerated video encoding and decoding
+ using the Intel Media SDK:
+
+ - oneVPL support (Intel oneAPI Video Processing Library)
+ - AV1 decoding support
+ - H.264 decoder now supports constrained-high and progressive-high
+ profiles
+ - H.264 encoder:
+ - more configuration options (properties):
+ "intra-refresh-type", "min-qp" , "max-qp", "p-pyramid",
+ "dblk-idc"
+ - H.265 encoder:
+ - can output main-still-picture profile
+ - now inserts HDR SEIs (mastering display colour volume and
+ content light level)
+ - more configuration options (properties):
+ "intra-refresh-type", "min-qp" , "max-qp", "p-pyramid",
+ "b-pyramid", "dblk-idc", "transform-skip"
+ - support for RGB 10bit format
+ - External bitrate control in encoders
+ - Video post proc element msdkvpp gained support for 12-bit pixel
+ formats P012_LE, Y212_LE and Y412_LE
+
+- nvh264sldec: interlaced stream support
+
+- openh264enc: support main, high, constrained-high and
+ progressive-high profiles
+
+- openjpeg: support for multithreaded decoding and encoding
+
+- rtspsrc: now supports IPv6 also for tunneled mode (RTSP-over-HTTP);
+ new "ignore-x-server-reply" property to ignore the
+ x-server-ip-address server header reply in case of HTTP tunneling,
+ as it is often broken.
+
+- souphttpsrc: Runtime compatibility support for libsoup2 and
+ libsoup3. libsoup3 is the latest major version of libsoup, but
+ libsoup2 and libsoup3 can’t co-exist in the same process because
+ there is no namespacing or versioning for GObject types. As a
+ result, it would be awkward if the GStreamer souphttpsrc plugin
+ linked to a specific version of libsoup, because it would only work
+ with applications that use the same version of libsoup. To make this
+ work, the soup plugin now tries to determine the libsoup version
+ used by the application (and its other dependencies) at runtime on
+ systems where GStreamer is linked dynamically. libsoup3 support is
+ still considered somewhat experimental at this point.
+
+- srtsrc, srtsink: add signals for the application to accept/reject
+ incoming connections
+
+- timeoverlay: new elapsed-running-time time mode which shows the
+ running time since the first running time (and each flush-stop).
+
+- udpsrc: new timestamping mode to retrieve packet receive timestamps
+ from the kernel via socket control messages (SO_TIMESTAMPNS) on
+ supported platforms
+
+- uritranscodebin: new setup-source and element-setup signals for
+ applications to configure elements used
+
+- v4l2codecs plugin gained support for 4x4 and 32x32 tile formats
+ enabling some platforms or direct renders. Important memory usage
+ improvement.
+
+- v4l2slh264dec now implements the final Linux uAPI as shipped on
+ Linux 5.11 and later.
+
+- valve: add "drop-mode" property and provide two new modes of
+ operation: in drop-mode=forward-sticky-events sticky events
+ (stream-start, segment, tags, caps, etc.) are forwarded downstream
+ even when dropping is enabled; drop-mode=transform-to-gap will in
+ addition also convert buffers into gap events when dropping is
+ enabled, which lets downstream elements know that time is advancing
+ and might allow for preroll in many scenarios. By default all events
+ and all buffers are dropped when dropping is enabled, which can
+ cause problems with caps negotiation not progressing or branches not
+ prerolling when dropping is enabled.
+
+- videocrop: support for many more pixel formats, e.g. planar YUV
+ formats with > 8bits and GBR* video formats; can now also accept
+ video not backed by system memory as long as downstream supports the
+ GstCropMeta
+
+- videotestsrc: new smpte-rp-219 pattern for SMPTE75 RP-219 conformant
+ color bars
+
+- vp8enc: finish support for temporal scalability: two new properties
+ ("temporal-scalability-layer-flags",
+ "temporal-scalability-layer-sync-flags") and a unit change on the
+ "temporal-scalability-target-bitrate" property (now expects bps);
+ also make temporal scalability details available to RTP payloaders
+ as buffer metadata.
+
+- vp9enc: new properties to tweak encoder performance:
+
+ - "aq-mode" to configure adaptive quantization modes
+ - "frame-parallel-decoding" to configure whether to create a
+ bitstream that reduces decoding dependencies between frames
+ which allows staged parallel processing of more than one video
+ frames in the decoder. (Defaults to TRUE)
+ - "row-mt", "tile-columns" and "tile-rows" so multithreading can
+ be enabled on a per-tile basis, instead of on a per tile-column
+ basis. In combination with the new "tile-rows" property, this
+ allows the encoder to make much better use of the available CPU
+ power.
+
+- vp9dec, vp9enc: add support for 10-bit 4:2:0 and 4:2:2 YUV, as well
+ as 8-bit 4:4:4
+
+- vp8enc, vp9enc now default to “good quality” for the deadline
+ property rather then “best quality”. Having the deadline set to best
+ quality causes the encoder to be absurdly slow, most real-life users
+ will prefer good-enough quality with better performance instead.
+
+- wpesrc:
+
+ - implement audio support: a new sometimes source pad will be
+ created for each audio stream created by the web engine.
+ - move wpesrc to wpevideosrc and add a wrapper bin wpesrc to also
+ support audio
+ - also handles web:// URIs now (same as cefsrc)
+ - post messages with the estimated load progress on the bus
-Plugin and library moves
+- x265enc: add negative DTS support, which means timestamps are now
+ offset by 1h same as with x264enc
+
+RTP Payloaders and Depayloaders
+
+- rtpisacpay, rtpisacdepay: new RTP payloader and depayloader for iSAC
+ audio codec
+
+- rtph264depay:
+
+ - new "request-keyframe" property to make the depayloader
+ automatically request a new keyframe from the sender on packet
+ loss, consistent with the new property on rtpvp8depay.
+ - new "wait-for-keyframe" property to make depayloader wait for a
+ new keyframe at the beginning and after packet loss (only
+ effective if the depayloader outputs AUs), consistent with the
+ existing property on rtpvp8depay.
+
+- rtpopuspay, rtpopusdepay: support libwebrtc-compatible multichannel
+ audio in addition to the previously supported multichannel audio
+ modes
+
+- rtpopuspay: add DTX (Discontinuous Transmission) support
+
+- rtpvp8depay: new "request-keyframe" property to make the depayloader
+ automatically request a new keyframe from the sender on packet loss.
+
+- rtpvp8pay: temporal scaling support
-- this section will be filled in in due course
+- rtpvp9depay: Improved SVC handling (aggregate all layers)
+
+RTP Infrastructure
+
+- rtpst2022-1-fecdec, rtpst2022-1-fecenc: new elements providing SMPTE
+ 2022-1 2-D Forward Error Correction. More details in Mathieu’s blog
+ post.
+
+- rtpreddec: BUNDLE support
+
+- rtpredenc, rtpulpfecenc: add support for Transport-wide Congestion
+ Control (TWCC)
+
+- rtpsession: new "twcc-feedback-interval" property to allow RTCP TWCC
+ reports to be scheduled on a timer instead of per marker-bit.
+
+Plugin and library moves
- There were no plugin moves or library moves in this cycle.
The following elements or plugins have been removed:
-- this section will be filled in in due course
-
- The ofa audio fingerprinting plugin has been removed. The MusicIP
database has been defunct for years so this plugin is likely neither
useful nor used by anyone.
Miscellaneous API additions
-- this section will be filled in in due course
+Core
+
+- gst_buffer_new_memdup() is a convenience function for the
+ widely-used gst_buffer_new_wrapped(g_memdup(data,size),size)
+ pattern.
+
+- gst_caps_features_new_single() creates a new single GstCapsFeatures,
+ avoiding the need to use the vararg function with NULL terminator
+ for simple cases.
+
+- gst_element_type_set_skip_documentation() can be used by plugins to
+ signal that certain elements should not be included in the GStreamer
+ plugin documentation. This is useful for plugins where elements are
+ registered dynamically based on hardware capabilities and/or where
+ the available plugins and properties vary from system to system.
+ This is used in the d3d11 plugin for example to ensure that only the
+ list of default elements is advertised in the documentation.
+
+- gst_type_find_suggest_empty_simple() is a new convenience function
+ for typefinders for cases where there’s only a media type and no
+ other fields.
+
+- New API to create elements and set properties at construction time,
+ which is not only convenient, but also allows GStreamer elements to
+ have construct-only properties: gst_element_factory_make_full(),
+ gst_element_factory_make_valist(),
+ gst_element_factory_make_with_properties(),
+ gst_element_factory_create_full(),
+ gst_element_factory_create_valist(),
+ gst_element_factory_create_with_properties().
+
+- GstSharedTaskPool: new “shared” task pool subclass with slightly
+ different default behaviour than the existing GstTaskPool which
+ would create unlimited number of threads for new tasks. The shared
+ taskpool creates up to N threads (default: 1) and then distributes
+ pending tasks to those threads round-robin style, and blocks if no
+ thread is available. It is possible to join tasks. This can be used
+ by plugins to implement simple multi-threaded processing and is used
+ for the new multi-threaded video conversion and compositing done in
+ GstVideoAggregator, videoconverter and compositor.
+
+Plugins Base Utils library
+
+- GstDiscoverer:
+
+ - gst_discoverer_container_info_get_tags() was added to retrieve
+ global/container tags (vs. per-stream tags). Per-Stream tags can
+ be retrieved via the existing
+ gst_discoverer_stream_info_get_tags().
+ gst_discoverer_info_get_tags(), which for many files returns a
+ confusing mix of stream and container tags, has been deprecated
+ in favour of the container/stream-specific functions.
+ - gst_discoverer_stream_info_get_stream_number() returns a unique
+ integer identifier for a given stream within the given
+ GstDiscoverer context. (If this matches the stream number inside
+ the container bitstream that’s by coincidence and not by
+ design.)
+
+- gst_pb_utils_get_caps_description_flags() can be used to query
+ whether certain caps represent a container, audio, video, image,
+ subtitles, tags, or something else. This only works for formats
+ known to GStreamer.
+
+- gst_pb_utils_get_file_extension_from_caps() returns a possible file
+ extension for given caps.
+
+- gst_codec_utils_h264_get_profile_flags_level(): Parses profile,
+ flags, and level from H264 AvcC codec_data. The format of H264 AVCC
+ extradata/sequence_header is documented in the ITU-T H.264
+ specification section 7.3.2.1.1 as well as in ISO/IEC 14496-15
+ section 5.3.3.1.2.
+
+- gst_codec_utils_caps_get_mime_codec() to convert caps to a RFC 6381
+ compatible MIME codec string codec. Useful for providing the codecs
+ field inside the Content-Type HTTP header for containerized formats,
+ such as mp4 or matroska.
+
+GStreamer OpenGL integration library and plugins
+
+- glcolorconvert: added suppport for converting the video formats
+ A420, AV12, BGR, BGRA, RGBP and BGRP.
+
+- Added support to GstGLBuffer for persistent buffer mappings where a
+ Pixel Buffer Object (PBO) can be mapped by both the CPU and the GPU.
+ This removes a memcpy() when uploading textures or vertices
+ particularly when software decoders (e.g. libav) are direct
+ rendering into our memory. Improves transfer performance
+ significantly. Requires OpenGL 4.4, GL_ARB_buffer_storage or
+ GL_EXT_buffer_storage
+
+- Added various helper functions for handling 4x4 matrices of affine
+ transformations as used by GstVideoAffineTransformationMeta.
+
+- Add support to GstGLContext for allowing the application to control
+ the config (EGLConfig, GLXConfig, etc) used when creating the OpenGL
+ context. This allows the ability to choose between RGB16 or RGB10A2
+ or RGBA8 back/front buffer configurations that were previously
+ hardcoded. GstGLContext also supports retrieving the configuration
+ it was created with or from an externally provide OpenGL context
+ handle. This infrastructure is also used to create a compatible
+ config from an application/externally provided OpenGL context in
+ order to improve compatibility with other OpenGL frameworks and GUI
+ toolkits. A new environment variable GST_GL_CONFIG was also added to
+ be able to request a specific configuration from the command line.
+ Note: different platforms will have different functionality
+ available.
+
+- Add support for choosing between EGL and WGL at runtime when running
+ on Windows. Previously this was a build-time switch. Allows use in
+ e.g. Gtk applications on Windows that target EGL/ANGLE without
+ recompiling GStreamer. gst_gl_display_new_with_type() can be used by
+ applications to choose a specific display type to use.
+
+- Build fixes to explicitly check for Broadcom-specific libraries on
+ older versions of the Raspberry Pi platform. The Broadcom OpenGL ES
+ and EGL libraries have different filenames. Using the vc4 Mesa
+ driver on the Raspberry Pi is not affected.
+
+- Added support to glupload and gldownload for transferring RGBA
+ buffers using the memory:NVMM available on the Nvidia Tegra family
+ of embedded devices.
+
+- Added support for choosing libOpenGL and libGLX as used in a GLVND
+ environment on unix-based platforms. This allows using desktop
+ OpenGL and EGL without pulling in any GLX symbols as would be
+ required with libGL.
+
+Video library
+
+- New raw video formats:
+
+ - AV12 (NV12 with alpha plane)
+ - RGBP and BGRP (planar RGB formats)
+ - ARGB64 variants with specified endianness instead of host
+ endianness:
+ - ARGB64_LE, ARGB64_BE
+ - RGBA64_BE, RGBA64_LE
+ - BGRA64_BE, BGRA64_LE
+ - ABGR64_BE, ABGR64_LE
+
+- gst_video_orientation_from_tag() is new convenience API to parse the
+ image orientation from a GstTagList.
+
+- GstVideoDecoder subframe support (see below)
+
+- GstVideoCodecState now also carries some HDR metadata
+
+- Ancillary video data: implement transform functions for AFD/Bar
+ metas, so they will be forwarded in more cases
+
+MPEG-TS library
+
+This library only handles section parsing and such, see above for
+changes to the actual mpegtsmux and mpegtsdemux elements.
+
+- many additions and improvements to SCTE-35 section parsing
+- new API for fetching extended descriptors:
+ gst_mpegts_find_descriptor_with_extension()
+- add support for SIT sections (Selection Information Tables)
+- expose event-from-section constructor gst_event_new_mpegts_section()
+- parse Audio Preselection Descriptor needed for Dolby AC-4
+
+GstWebRTC library + webrtcbin
+
+- Change the way in which sink pads and transceivers are matched
+ together to support easier usage. If a pad is created without a
+ specific index (i.e. using sink_%u as the pad template), then an
+ available compatible transceiver will be searched for. If a specific
+ index is requested (i.e. sink_1) then if a transceiver for that
+ m-line already exists, that transceiver must match the new sink pad
+ request. If there is no transceiver available in either scenario, a
+ new transceiver is created. If a mixture of both sink_1 and sink_%u
+ requests result in an impossible situation, an error will be
+ produced at pad request time or from create offer/answer.
+
+- webrtcbin now uses regular ICE nomination instead of libnice’s
+ default of aggressive ICE nomination. Regular ICE nomination is the
+ default recommended by various relevant standards and improves
+ connectivity in specific network scenarios.
+
+- Add support for limiting the port range used for RTP with the
+ addition of the min-rtp-port and max-rtp-port properties on the ICE
+ object.
+
+- Expose the SCTP transport as a property on webrtcbin to more closely
+ match the WebRTC specification.
+
+- Added support for taking into account the data channel transport
+ state when determining the value of the "connection-state" property.
+ Previous versions of the WebRTC spec did not include the data
+ channel state when computing this value.
+
+- Add configuration for choosing the size of the underlying sockets
+ used for transporting media data
+
+- Always advertise support for the transport-cc RTCP feedback protocol
+ as rtpbin supports it. For full support, the configured caps (input
+ or through codec-preferences) need to include the relevant RTP
+ header extension.
+
+- Numerous fixes to caps and media handling to fail-fast when an
+ incompatible situation is detected.
+
+- Improved support for attaching the required media after a remote
+ offer has been set.
+
+- Add support for dynamically changing the amount of FEC used for a
+ particular stream.
+
+- webrtcbin now stops further SDP processing at the first error it
+ encounters.
+
+- Completed support for either local or the remote closing a data
+ channel.
+
+- Various fixes when performing BUNDLEing of the media streams in
+ relation to RTX and FEC usage.
+
+- Add support for writing out QoS DSCP marking on outgoing packets to
+ improve reliability in some network scenarios.
+
+- Improvements to the statistics returned by the get-stats signal
+ including the addition of the raw statistics from the internal
+ RTPSource, the TWCC stats when available.
+
+- The webrtc library does not expose any objects anymore with public
+ fields. Instead properties have been added to replace that
+ functionality. If you are accessing such fields in your application,
+ switch to the corresponding properties.
+
+GstCodecs and Video Parsers
+
+- Support for render delays to improve throughput across all CODECs
+ (used with NVDEC and V4L2).
+- lots of improvements to parsers and the codec parsing decoder base
+ classes (H264, H265, VP8, VP9, AV1, MPEG-2) used for various
+ hardware-accelerate decoder APIs.
+
+Bindings support
+
+- gst_allocation_params_new() allocates a GstAllocationParams struct
+ on the heap. This should only be used by bindings (and freed via
+ gst_allocation_params_free() then). In C code you would allocate
+ this on the stack and only init it in place.
+
+- gst_debug_log_literal() can be used to log a string to the debug log
+ without going through any printf format expansion and associated
+ overhead. This is mostly useful for bindings such as the Rust
+ bindings which may have done their own formatting already .
+
+- Provide non-inlined versions of refcounting APIs for various
+ GStreamer mini objects, so that they can be consumed by bindings
+ (e.g. gstreamer-sharp): gst_buffer_ref, gst_buffer_unref,
+ gst_clear_buffer, gst_buffer_copy, gst_buffer_replace,
+ gst_buffer_list_ref, gst_buffer_list_unref, gst_clear_buffer_list,
+ gst_buffer_list_copy, gst_buffer_list_replace, gst_buffer_list_take,
+ gst_caps_ref, gst_caps_unref, gst_clear_caps, gst_caps_replace,
+ gst_caps_take, gst_context_ref, gst_context_unref, gst_context_copy,
+ gst_context_replace, gst_event_replace, gst_event_steal,
+ gst_event_take, gst_event_ref, gst_event_unref, gst_clear_event,
+ gst_event_copy, gst_memory_ref, gst_memory_unref, gst_message_ref,
+ gst_message_unref, gst_clear_message, gst_message_copy,
+ gst_message_replace, gst_message_take, gst_promise_ref,
+ gst_promise_unref, gst_query_ref, gst_query_unref, gst_clear_query,
+ gst_query_copy, gst_query_replace, gst_query_take, gst_sample_ref,
+ gst_sample_unref, gst_sample_copy, gst_tag_list_ref,
+ gst_tag_list_unref, gst_clear_tag_list, gst_tag_list_replace,
+ gst_tag_list_take, gst_uri_copy, gst_uri_ref, gst_uri_unref,
+ gst_clear_uri.
+
+- expose a GType for GstMiniObject
+
+- gst_device_provider_probe() now returns non-floating device object
+
+API Deprecations
+
+- gst_element_get_request_pad() has been deprecated in favour of the
+ newly-added gst_element_request_pad_simple() which does the exact
+ same thing but has a less confusing name that hopefully makes clear
+ that the function request a new pad rather than just retrieves an
+ already-existing request pad.
+
+- gst_discoverer_info_get_tags(), which for many files returns a
+ confusing mix of stream and container tags, has been deprecated in
+ favour of the container-specific and stream-specific functions,
+ gst_discoverer_container_info_get_tags() and
+ gst_discoverer_stream_info_get_tags().
+
+- gst_video_sink_center_rect() was deprecated in favour of the more
+ generic newly-added gst_video_center_rect().
+
+- The GST_MEMORY_FLAG_NO_SHARE flag has been deprecated, as it tends
+ to cause problems and prevents sub-buffering. If pooling or lifetime
+ tracking is required, memories should be allocated through a custom
+ GstAllocator instead of relying on the lifetime of the buffers the
+ memories were originally attached to, which is fragile anyway.
+
+- The GstPlayer high-level playback library is being replaced with the
+ new GstPlay library (see above). GstPlayer should be considered
+ deprecated at this point and will be marked as such in the next
+ development cycle. Applications should be ported to GstPlay.
+
+- Gstreamer Editing Services: ges_video_transition_set_border(),
+ ges_video_transition_get_border()
+ ges_video_transition_set_inverted()
+ ges_video_transition_is_inverted() have been deprecated, use
+ ges_timeline_element_set_children_properties() instead.
Miscellaneous performance, latency and memory optimisations
-- this section will be filled in in due course
+More video conversion fast paths
-Miscellaneous other changes and enhancements
+- v210 ↔ I420, YV12, Y42B, UYVY and YUY2
+- A420 → RGB
+
+Less jitter when waiting on the system clock
+
+- Better system clock wait accuracy, less jitter: where available,
+ clock_nanosleep is used for higher accuracy for waits below 500
+ usecs, and waits below 2ms will first use the regular waiting system
+ and then clock_nanosleep for the remainder. The various wait
+ implementation have a latency ranging from 50 to 500+ microseconds.
+ While this is not a major issue when dealing with a low number of
+ waits per second (for ex: video), it does introduce a non-negligible
+ jitter for synchronisation of higher packet rate systems.
+
+Video decoder subframe support
-- this section will be filled in in due course
+- The GstVideoDecoder base class gained API to process input at the
+ sub-frame level. That way video decoders can start decoding slices
+ before they have received the full input frame in its entirety (to
+ the extent this is supported by the codec, of course). This helps
+ with CPU utilisation and reduces latency.
-Tracing framework and debugging improvements
+- This functionality is now being used in the OpenJPEG JPEG 2000
+ decoder, the FFmpeg H.264 decoder (in case of NAL-aligned input) and
+ the OpenMAX H.264/H.265 decoders (in case of NAL-aligned input).
-- this section will be filled in in due course
+Miscellaneous other changes and enhancements
+
+- GstDeviceMonitor no longer fails to start just because one of the
+ device providers failed to start. That could happen for example on
+ systems where the pulseaudio device provider is installed, but
+ pulseaudio isn’t actually running but ALSA is used for audio
+ instead. In the same vein the device monitor now keeps track of
+ which providers have been started (via the new
+ gst_device_provider_is_started()) and only stops actually running
+ device providers when stopping the device monitor.
+
+- On embedded systems it can be useful to create a registry that can
+ be shared and read by multiple processes running as different users.
+ It is now possible to set the new GST_REGISTRY_MODE environment
+ variable to specify the file mode for the registry file, which by
+ default is set to be only user readable/writable.
+
+- GstNetClientClock will signal lost sync in case the remote time
+ resets (e.g. because device power cycles), by emitting the “synced”
+ signal with synced=FALSE parameter, so applications can take action.
+
+- gst_value_deserialize_with_pspec() allows deserialization with a
+ hint for what the target GType should be. This allows for example
+ passing arrays of flags through the command line or
+ gst_util_set_object_arg(), eg: foo="<bar,bar+baz>".
+
+- It’s now allowed to create an empty GstVideoOverlayComposition
+ without any rectangles by passing a NULL rectangle to
+ gst_video_overlay_composition_new(). This is useful for bindings and
+ simplifies application code in some places.
+
+Tracing framework, debugging and testing improvements
+
+- New factories tracer to list loaded elements (and other plugin
+ features). This can be useful to collect a list of elements needed
+ for an application, which then in turn can be used to create a
+ tailored minimal GStreamer build that contains just the elements
+ needed and nothing else.
+- New plugin-feature-loaded tracing hook for use by tracers like the
+ new factories tracer
+
+- GstHarness: Add gst_harness_set_live() so that harnesses can be set
+ to non-live and return is-live=false in latency queries if needed.
+ Default behaviour is to always return is-live=true in latency
+ queries.
+
+- navseek: new "hold-eos" property. When enabled, the element will
+ hold back an EOS event until the next keystroke (via navigation
+ events). This can be used to keep a video sink showing the last
+ frame of a video pipeline until a key is pressed instead of tearing
+ it down immediately on EOS.
+
+- New fakeaudiosink element: mimics an audio sink and can be used for
+ testing and CI pipelines on systems where no audio system is
+ installed or running. It differs from fakesink in that it only
+ support audio caps and syncs to the clock by default like a normal
+ audio sink. It also implements the GstStreamVolume interface like
+ most audio sinks do.
+
+- New videocodectestsink element for video codec conformance testing:
+ Calculates MD5 checksums for video frames and skips any padding
+ whilst doing so. Can optionally also write back the video data with
+ padding removed into a file for easy byte-by-byte comparison with
+ reference data.
Tools
-- this section will be filled in in due course
+gst-inspect-1.0
+
+- Can sort the list of plugins by passing --sort=name as command line
+ option
+
+gst-launch-1.0
+
+- will now error out on top-level properties that don’t exist and
+ which were silently ignored before
+- On Windows the high-resolution clock is enabled now, which provides
+ better clock and timer performance on Windows (see Windows section
+ below for more details).
+
+gst-play-1.0
+
+- New --start-position command line argument to start playback from
+ the specified position
+- Audio can be muted/unmuted in interactive mode by pressing the m
+ key.
+- On Windows the high-resolution clock is enabled now (see Windows
+ section below for more details)
+
+gst-device-monitor-1.0
+
+- New --include-hidden command line argument to also show “hidden”
+ device providers
+
+ges-launch-1.0
+
+- New interactive mode that allows seeking and such. Can be disabled
+ by passing the --no-interactive argument on the command line.
+- Option to forward tags
+- Allow using an existing clip to determine the rendering format (both
+ topology and profile) via new --profile-from command line argument.
GStreamer RTSP server
-- this section will be filled in in due course
+- GstRTSPMediaFactory gained API to disable RTCP
+ (gst_rtsp_media_factory_set_enable_rtcp(), "enable-rtcp" property).
+ Previously RTCP was always allowed for all RTSP medias. With this
+ change it is possible to disable RTCP completely, no matter if the
+ client wants to do RTCP or not.
+
+- Make a mount point of / work correctly. While not allowed by the
+ RTSP 2 spec, the RTSP 1 spec is silent on this and it is used in the
+ wild. It is now possible to use / as a mount path in
+ gst-rtsp-server, e.g. rtsp://example.com/ would work with this now.
+ Note that query/fragment parts of the URI are not necessarily
+ correctly handled, and behaviour will differ between various
+ client/server implementations; so use it if you must but don’t bug
+ us if it doesn’t work with third party clients as you’d hoped.
+
+- multithreading fixes (races, refcounting issues, deadlocks)
+
+- ONVIF audio backchannel fixes
+
+- ONVIF trick mode optimisations
+
+- rtspclientsink: new "update-sdp" signal that allows updating the SDP
+ before sending it to the server via ANNOUNCE. This can be used to
+ add additional metadata to the SDP, for example. The order and
+ number of medias must not be changed, however.
GStreamer VAAPI
-- this section will be filled in in due course
+- new AV1 decoder element (vaapiav1dec)
+
+- H264 decoder: handle stereoscopic 3D video with frame packing
+ arrangement SEI messages
+
+- H265 encoder: added Screen Content Coding extensions support
+
+- H265 decoder: gained MAIN_444_12 profile support (decoded to
+ Y412_LE), and 4:2:2 12-bits support (decoded to Y212_LE)
+
+- vaapipostproc: gained BT2020 color standard support
+
+- vaapidecode: now generates caps templates dynamically at runtime in
+ order to advertise actually supported caps instead of all
+ theoretically supported caps.
+
+- GST_VAAPI_DRM_DEVICE environment variable to force a specified DRM
+ device when a DRM display is used. It is ignored when other types of
+ displays are used. By default /dev/dri/renderD128 is used for DRM
+ display.
GStreamer OMX
-- this section will be filled in in due course
+- subframe support in H.264/H.265 decoders
GStreamer Editing Services and NLE
-- this section will be filled in in due course
+- framepositioner: new "operator" property to access blending modes in
+ the compositor
+- timeline: Implement snapping to markers
+- smart-mixer: Add support for d3d11compositor and glvideomixer
+- titleclip: add "draw-shadow" child property
+- ges:// URI support to define a timeline from a description.
+- command-line-formatter
+ - Add track management to timeline description
+ - Add keyframe support
+- ges-launch-1.0:
+ - Add an interactive mode where we can seek etc…
+ - Add option to forward tags
+ - Allow using an existing clip to determine the rendering format
+ (both topology and profile) via new --profile-from command line
+ argument.
+- Fix static build
GStreamer validate
-- this section will be filled in in due course
+- report: Add a way to force backtraces on reports even if not a
+ critical issue (GST_VALIDATE_ISSUE_FLAGS_FORCE_BACKTRACE)
+- Add a flag to gst_validate_replace_variables_in_string() allow
+ defining how to resolve variables in structs
+- Add gst_validate_bin_monitor_get_scenario() to get the bin monitor
+ scenario, which is useful for applications that use Validate
+ directly.
+- Add an expected-values parameter to wait, message-type=XX allowing
+ more precise filtering of the message we are waiting for.
+- Add config file support: each test can now use a config file for the
+ given media file used to test.
+- Add support to check properties of object properties
+- scenario: Add an "action-done" signal to signal when an action is
+ done
+- scenario: Add a "run-command" action type
+- scenario: Allow forcing running action on idle from scenario file
+- scenario: Allow iterating over arrays in foreach
+- scenario: Rename ‘interlaced’ action to ‘non-blocking’
+- scenario: Add a non-blocking flag to the wait signal
GStreamer Python Bindings
-- this section will be filled in in due course
+- Fixes for Python 3.10
+- Various build fixes
+- at least one known breaking change caused by g-i annotation changes
+ (see below)
GStreamer C# Bindings
-- this section will be filled in in due course
+- Fix GstDebugGraphDetails enum
+- Updated to latests GtkSharp
+- Updated to include GStreamer 1.20 API
GStreamer Rust Bindings and Rust Plugins
-The GStreamer Rust bindings are released separately with a different
-release cadence that’s tied to gtk-rs, but the latest release has
-already been updated for the upcoming new GStreamer 1.20 API.
-
-gst-plugins-rs, the module containing GStreamer plugins written in Rust,
-has also seen lots of activity with many new elements and plugins.
-
-What follows is a list of elements and plugins available in
-gst-plugins-rs, so people don’t miss out on all those potentially useful
-elements that have no C equivalent.
-
-- FIXME: add new elements
-
-Rust audio plugins
-
-- audiornnoise: New element for audio denoising which implements the
- noise removal algorithm of the Xiph RNNoise library, in Rust
-- rsaudioecho: Port of the audioecho element from gst-plugins-good
- rsaudioloudnorm: Live audio loudness normalization element based on
- the FFmpeg af_loudnorm filter
-- claxondec: FLAC lossless audio codec decoder element based on the
- pure-Rust claxon implementation
-- csoundfilter: Audio filter that can use any filter defined via the
- Csound audio programming language
-- lewtondec: Vorbis audio decoder element based on the pure-Rust
- lewton implementation
-
-Rust video plugins
-
-- cdgdec/cdgparse: Decoder and parser for the CD+G video codec based
- on a pure-Rust CD+G implementation, used for example by karaoke CDs
-- cea608overlay: CEA-608 Closed Captions overlay element
-- cea608tott: CEA-608 Closed Captions to timed-text (e.g. VTT or SRT
- subtitles) converter
-- tttocea608: CEA-608 Closed Captions from timed-text converter
-- mccenc/mccparse: MacCaption Closed Caption format encoder and parser
-- sccenc/sccparse: Scenarist Closed Caption format encoder and parser
-- dav1dec: AV1 video decoder based on the dav1d decoder implementation
- by the VLC project
-- rav1enc: AV1 video encoder based on the fast and pure-Rust rav1e
- encoder implementation
-- rsflvdemux: Alternative to the flvdemux FLV demuxer element from
- gst-plugins-good, not feature-equivalent yet
-- rsgifenc/rspngenc: GIF/PNG encoder elements based on the pure-Rust
- implementations by the image-rs project
-
-Rust text plugins
-
-- textwrap: Element for line-wrapping timed text (e.g. subtitles) for
- better screen-fitting, including hyphenation support for some
- languages
-
-Rust network plugins
-
-- reqwesthttpsrc: HTTP(S) source element based on the Rust
- reqwest/hyper HTTP implementations and almost feature-equivalent
- with the main GStreamer HTTP source souphttpsrc
-- s3src/s3sink: Source/sink element for the Amazon S3 cloud storage
-- awstranscriber: Live audio to timed text transcription element using
- the Amazon AWS Transcribe API
-
-Generic Rust plugins
-
-- sodiumencrypter/sodiumdecrypter: Encryption/decryption element based
- on libsodium/NaCl
-- togglerecord: Recording element that allows to pause/resume
- recordings easily and considers keyframe boundaries
-- fallbackswitch/fallbacksrc: Elements for handling potentially
- failing (network) sources, restarting them on errors/timeout and
- showing a fallback stream instead
-- threadshare: Set of elements that provide alternatives for various
- existing GStreamer elements but allow to share the streaming threads
- between each other to reduce the number of threads
-- rsfilesrc/rsfilesink: File source/sink elements as replacements for
- the existing filesrc/filesink elements
+- The GStreamer Rust bindings are released separately with a different
+ release cadence that’s tied to gtk-rs, but the latest release has
+ already been updated for the upcoming new GStreamer 1.20 API (v1_20
+ feature).
+
+- gst-plugins-rs, the module containing GStreamer plugins written in
+ Rust, has also seen lots of activity with many new elements and
+ plugins. See the New Elements section above for a list of new Rust
+ elements.
Build and Dependencies
- Meson 0.59 or newer is required to build GStreamer now.
-- FIXME: this section will be filled in in due course
+- The GLib requirement has been bumped to GLib 2.56 or newer (from
+ March 2018).
+
+- The wpe plugin now requires wpe >= 2.28 and wpebackend-fdo >= 1.8
Explicit opt-in required for build of certain plugins with (A)GPL dependencies
See Building plugins with (A)GPL-licensed dependencies for more details
and a non-exhaustive list of plugins affected.
-gst-build: replaced by Monorepo
+gst-build: replaced by mono repository
-- this section will be filled in in due course
-
-- FIXME: describe + link to Monorepo FAQ
+See mono repository section above and the GStreamer mono repository FAQ.
Cerbero
on platforms where dependencies are not readily available, such as
Windows, Android, iOS and macOS.
-General improvements
+General Cerbero improvements
-- this section will be filled in in due course
+- Plugin removed: libvisual
+- New plugins: rtpmanagerbad and rist
-macOS / iOS
+macOS / iOS specific Cerbero improvements
-- this section will be filled in in due course
+- XCode 12 support
+- macOS OS release support is now future-proof, similar to iOS
+- macOS Apple Silicon (ARM64) cross-compile support has been added
+- macOS Apple Silicon (ARM64) native support is currently experimental
-Windows
+Windows specific Cerbero improvements
-- this section will be filled in in due course
+- Visual Studio 2022 support has been added
+- bootstrap is faster since it requires building fewer build-tools
+ recipes on Windows
+- package is faster due to better scheduling of recipe stages and
+ elimination of unnecessary autotools regeneration
+- The following plugins are no longer built on Windows:
+ - a52dec (another decoder is still available in libav)
+ - dvdread
+ - resindvd
Windows MSI installer
-- this section will be filled in in due course
+- no major changes
-Linux
+Linux specific Cerbero improvements
-- this section will be filled in in due course
+- Fedora, Debian OS release support is now more future-proof
+- Amazon Linux 2 support has been added
-Android
+Android specific Cerbero improvements
-- this section will be filled in in due course
+- no major changes
Platform-specific changes and improvements
Android
-- this section will be filled in in due course
+- No major changes
macOS and iOS
-- this section will be filled in in due course
+- applemedia: add ProRes support to vtenc and vtdec
Windows
-- this section will be filled in in due course
+- On Windows the high-resolution clock is enabled now in the
+ gst-launch-1.0 and gst-play-1.0 command line tools, which provides
+ better clock and timer performance on Windows, at the cost of higher
+ power consumption. By default, without the high-resolution clock
+ enabled, the timer precision on Windows is system-dependent and may
+ be as bad as 15ms which is not good enough for many multimedia
+ applications. Developers may want to do the same in their Windows
+ applications if they think it’s a good idea for their application
+ use case, and depending on the Windows version they target. This is
+ not done automatically by GStreamer because on older Windows
+ versions (pre-Windows 10) this affects a global Windows setting and
+ also there’s a power consumption vs. performance trade-off that may
+ differ from application to application.
+
+- dxgiscreencapsrc now supports resolution changes
+
+- The wasapi2 audio plugin was rewritten and now has a higher rank
+ than the old wasapi plugin since it has a number of additional
+ features such as automatic stream routing, and no
+ known-but-hard-to-fix issues. The plugin is always built if the
+ Windows 10 SDK is available now.
+
+- The wasapi device providers now detect and notify dynamic device
+ additions/removals
+
+- d3d11screencapturesrc: new desktop capture element, including
+ GstDeviceProvider implementation to enumerate/select target monitors
+ for capture.
+
+- Direct3D11/DXVA decoder now supports AV1 and MPEG2 codecs
+ (d3d11av1dec, d3d11mpeg2dec)
+
+- VP9 decoding got more reliable and stable thanks to a newly written
+ codec parser
+
+- Support for decoding interlaced H.264/AVC streams
+
+- Hardware-accelerated video deinterlacing (d3d11deinterlace) and
+ video mixing (d3d11compositor)
+
+- Video mixing with the Direct3D11 API (d3d11compositor)
+
+- MediaFoundation API based hardware encoders gained the ability to
+ receive Direct3D11 textures as an input
+
+- Seungha’s blog post “GStreamer ❤ Windows: A primer on the cool stuff
+ you’ll find in the 1.20 release” describes many of the
+ Windows-related improvements in more detail
Linux
-- this section will be filled in in due course
+- bluez: LDAC Bluetooth audio codec support in a2dpsink and avdtpsink,
+ as well as an LDAC RTP payloader (rtpldacpay) and an LDAC audio
+ encoder (ldacenc)
+
+- kmssink: gained support for NV24, NV61, RGB16/BGR16 formats;
+ auto-detect NVIDIA Tegra driver
Documentation improvements
-- this section will be filled in in due course
+- hardware-accelerated GPU plugins will now no longer always list all
+ the element variants for all available GPUs, since those are
+ system-dependent and it’s confusing for users to see those in the
+ documentation just because the GStreamer developer who generated the
+ docs had multiple GPUs to play with at the time. Instead just show
+ the default elements.
-Possibly Breaking Changes
+Possibly Breaking and Other Noteworthy Behavioural Changes
+
+- gst_parse_launch(), gst_parse_bin_from_description() and friends
+ will now error out when setting properties that don’t exist on
+ top-level bins. They were silently ignored before.
+
+- The GstWebRTC library does not expose any objects anymore with
+ public fields. Instead properties have been added to replace that
+ functionality. If you are accessing such fields in your application,
+ switch to the corresponding properties.
-- FIXME: this section will be filled in in due course
-- MPEG-TS SCTE-35 API changes (FIXME: flesh out)
-- gst_parse_launch() and friends now error out on non-existing
- properties on top-level bins where they would silently fail and
- ignore those before.
- playbin and uridecodebin now emit the source-setup signal before the
element is added to the bin and linked so that the source element is
already configured before any scheduling query comes in, which is
- useful for elements such as appsrc or giostreamsrc. (Merge Request)
+ useful for elements such as appsrc or giostreamsrc.
+
+- The source element inside urisourcebin (used inside uridecodebin3
+ which is used inside playbin3) is no longer called "source". This
+ shouldn’t affect anyone hopefully, because there’s a "setup-source"
+ signal to configure the source element and no one should rely on
+ names of internal elements anyway.
+
+- The vp8enc element now expects bps (bits per second) for the
+ "temporal-scalability-target-bitrate" property, which is consistent
+ with the "target-bitrate" property. Since additional configuration
+ is required with modern libvpx to make temporal scaling work anyway,
+ chances are that very few people will have been using this property
+
+- vp8enc and vp9enc now default to “good quality” for the "deadline"
+ property rather then “best quality”. Having the deadline set to best
+ quality causes the encoder to be absurdly slow, most real-life users
+ will want the good quality tradeoff instead.
+
+- The experimental GstTranscoder library API in gst-plugins-bad was
+ changed from a GObject signal-based notification mechanism to a
+ GstBus/message-based mechanism akin to GstPlayer/GstPlay.
+
+- MPEG-TS SCTE-35 API: semantic change for SCTE-35 splice commands:
+ timestamps passed by the application should be in running time now,
+ since users of the API can’t really be expected to predict the local
+ PTS of the muxer.
+
+- The GstContext used by souphttpsrc to share the session between
+ multiple element instances has changed. Previously it provided
+ direct access to the internal SoupSession object, now it only
+ provides access to an opaque, internal type. This change is
+ necessary because SoupSession is not thread-safe at all and can’t be
+ shared safely between arbitrary external code and souphttpsrc.
+
+- Python bindings: GObject-introspection related Annotation fixes have
+ led to a case of a GstVideo.VideoInfo-related function signature
+ changing in the Python bindings (possibly one or two other cases
+ too). This is for a function that should never have been exposed in
+ the first place though, so the bindings are being updated to throw
+ an exception in that case, and the correct replacement API has been
+ added in form of an override.
Known Issues
-- this section will be filled in in due course
-
-- There are a couple of known WebRTC-related regressions/blockers:
-
- - webrtc: DTLS setup with Chrome is broken
- - webrtcbin: First keyframe is usually lost
+- nothing in particular at this point (but also see possibly breaking
+ changes section above)
Contributors
-- this section will be filled in in due course
+Aaron Boxer, Adam Leppky, Adam Williamson, Alba Mendez, Alejandro
+González, Aleksandr Slobodeniuk, Alexander Vandenbulcke, Alex Ashley,
+Alicia Boya García, Andika Triwidada, Andoni Morales Alastruey, Andrew
+Wesie, Andrey Moiseev, Antonio Ospite, Antonio Rojas, Arthur Crippa
+Búrigo, Arun Raghavan, Ashley Brighthope, Axel Kellermann, Baek, Bastien
+Nocera, Bastien Reboulet, Benjamin Gaignard, Bing Song, Binh Truong,
+Biswapriyo Nath, Brad Hards, Brad Smith, Brady J. Garvin, Branko
+Subasic, Camilo Celis Guzman, Chris Bass, ChrisDuncanAnyvision, Chris
+White, Corentin Damman, Daniel Almeida, Daniel Knobe, Daniel Stone,
+david, David Fernandez, David Keijser, David Phung, Devarsh Thakkar,
+Dinesh Manajipet, Dmitry Samoylov, Dmitry Shusharin, Dominique Martinet,
+Doug Nazar, Ederson de Souza, Edward Hervey, Emmanuel Gil Peyrot,
+Enrique Ocaña González, Ezequiel Garcia, Fabian Orccon, Fabrice
+Fontaine, Fernando Jimenez Moreno, Florian Karydes, Francisco Javier
+Velázquez-García, François Laignel, Frederich Munch, Fredrik Pålsson,
+George Kiagiadakis, Georg Lippitsch, Göran Jönsson, Guido Günther,
+Guillaume Desmottes, Guiqin Zou, Haakon Sporsheim, Haelwenn (lanodan)
+Monnier, Haihao Xiang, Haihua Hu, Havard Graff, He Junyan, Helmut
+Januschka, Henry Wilkes, Hosang Lee, Hou Qi, Ignacio Casal Quinteiro,
+Igor Kovalenko, Ilya Kreymer, Imanol Fernandez, Jacek Tomaszewski, Jade
+Macho, Jakub Adam, Jakub Janků, Jan Alexander Steffens (heftig), Jan
+Schmidt, Jason Carrete, Jason Pereira, Jay Douglass, Jeongki Kim, Jérôme
+Laheurte, Jimmi Holst Christensen, Johan Sternerup, John Hassell, John
+Lindgren, John-Mark Bell, Jonathan Matthew, Jordan Petridis, Jose
+Quaresma, Julian Bouzas, Julien, Kai Uwe Broulik, Kasper Steensig
+Jensen, Kellermann Axel, Kevin Song, Khem Raj, Knut Inge Hvidsten, Knut
+Saastad, Kristofer Björkström, Lars Lundqvist, Lawrence Troup, Lim Siew
+Hoon, Lucas Stach, Ludvig Rappe, Luis Paulo Fernandes de Barros, Luke
+Yelavich, Mads Buvik Sandvei, Marc Leeman, Marco Felsch, Marek Vasut,
+Marian Cichy, Marijn Suijten, Marius Vlad, Markus Ebner, Mart Raudsepp,
+Matej Knopp, Mathieu Duponchelle, Matthew Waters, Matthieu De Beule,
+Mengkejiergeli Ba, Michael de Gans, Michael Olbrich, Michael Tretter,
+Michal Dzik, Miguel Paris, Mikhail Fludkov, mkba, Nazar Mokrynskyi,
+Nicholas Jackson, Nicola Murino, Nicolas Dufresne, Niklas Hambüchen,
+Nikolay Sivov, Nirbheek Chauhan, Olivier Blin, Olivier Crete, Olivier
+Crête, Paul Goulpié, Per Förlin, Peter Boba, P H, Philippe Normand,
+Philipp Zabel, Pieter Willem Jordaan, Piotrek Brzeziński, Rafał
+Dzięgiel, Rafostar, raghavendra, Raghavendra, Raju Babannavar, Raleigh
+Littles III, Randy Li, Randy Li (ayaka), Ratchanan Srirattanamet, Raul
+Tambre, reed.lawrence, Ricky Tang, Robert Rosengren, Robert Swain, Robin
+Burchell, Roman Sivriver, R S Nikhil Krishna, Ruben Gonzalez, Ruslan
+Khamidullin, Sanchayan Maity, Scott Moreau, Sebastian Dröge, Sergei
+Kovalev, Seungha Yang, Sid Sethupathi, sohwan.park, Sonny Piers, Staz M,
+Stefan Brüns, Stéphane Cerveau, Stephan Hesse, Stian Selnes, Stirling
+Westrup, Théo MAILLART, Thibault Saunier, Tim, Timo Wischer, Tim-Philipp
+Müller, Tim Schneider, Tobias Ronge, Tom Schoonjans, Tulio Beloqui,
+tyler-aicradle, U. Artie Eoff, Ung, Val Doroshchuk, VaL Doroshchuk,
+Víctor Manuel Jáquez Leal, Vivek R, Vivia Nikolaidou, Vivienne
+Watermeier, Vladimir Menshakov, Will Miller, Wim Taymans, Xabier
+Rodriguez Calvar, Xavier Claessens, Xℹ Ruoyao, Yacine Bandou, Yinhang
+Liu, youngh.lee, youngsoo.lee, yychao, Zebediah Figura, Zhang yuankun,
+Zhang Yuankun, Zhao, Zhao Zhili, , Aleksandar Topic, Antonio Ospite,
+Bastien Nocera, Benjamin Gaignard, Brad Hards, Carlos Falgueras García,
+Célestin Marot, Corentin Damman, Corentin Noël, Daniel Almeida, Daniel
+Knobe, Danny Smith, Dave Piché, Dmitry Osipenko, Fabrice Fontaine,
+fjmax, Florian Zwoch, Guillaume Desmottes, Haihua Hu, Heinrich Kruger,
+He Junyan, Jakub Adam, James Cowgill, Jan Alexander Steffens (heftig),
+Jean Felder, Jeongki Kim, Jiri Uncovsky, Joe Todd, Jordan Petridis,
+Krystian Wojtas, Marc-André Lureau, Marcin Kolny, Marc Leeman, Mark
+Nauwelaerts, Martin Reboredo, Mathieu Duponchelle, Matthew Waters,
+Mengkejiergeli Ba, Michael Gruner, Nicolas Dufresne, Nirbheek Chauhan,
+Olivier Crête, Philippe Normand, Rafał Dzięgiel, Ralf Sippl, Robert
+Mader, Sanchayan Maity, Sangchul Lee, Sebastian Dröge, Seungha Yang,
+Stéphane Cerveau, Teh Yule Kim, Thibault Saunier, Thomas Klausner, Timo
+Wischer, Tim-Philipp Müller, Tobias Reineke, Tomasz Andrzejak, Trung Do,
+Tyler Compton, Ung, Víctor Manuel Jáquez Leal, Vivia Nikolaidou, Wim
+Taymans, wngecn, Wonchul Lee, wuchang li, Xavier Claessens, Xi Ruoyao,
+Yoshiharu Hirose, Zhao,
… and many others who have contributed bug reports, translations, sent
suggestions or helped testing.
1.20.0
-1.20.0 is scheduled to be released around October/November 2021.
+1.20.0 is scheduled to be released around early February 2022.
Schedule for 1.22
unstable development version leading up to the stable 1.22 release. The
development of 1.21/1.22 will happen in the git main branch.
-The plan for the 1.22 development cycle is yet to be confirmed.
+The plan for the 1.22 development cycle is yet to be confirmed. Assuming
+no major project-wide reorganisations in the 1.22 cycle we might try and
+aim for a release around August 2022.
1.22 will be backwards-compatible to the stable 1.20, 1.18, 1.16, 1.14,
1.12, 1.10, 1.8, 1.6, 1.4, 1.2 and 1.0 release series.
------------------------------------------------------------------------
These release notes have been prepared by Tim-Philipp Müller with
-contributions from …
+contributions from Matthew Waters, Nicolas Dufresne, Nirbheek Chauhan,
+Sebastian Dröge and Seungha Yang.
License: CC BY-SA 4.0
-This is GStreamer gst-libav 1.19.3.
+This is GStreamer gst-libav 1.19.90.
GStreamer 1.19 is the development branch leading up to the next major
stable version which will be 1.20.
<release>
<Version>
+ <revision>1.19.90</revision>
+ <branch>main</branch>
+ <name></name>
+ <created>2022-01-28</created>
+ <file-release rdf:resource="https://gstreamer.freedesktop.org/src/gst-libav/gst-libav-1.19.90.tar.xz" />
+ </Version>
+ </release>
+
+ <release>
+ <Version>
<revision>1.19.3</revision>
<branch>main</branch>
<name></name>
project('gst-libav', 'c', 'cpp',
- version : '1.19.3.1',
+ version : '1.19.90',
meson_version : '>= 0.59',
default_options : [ 'warning_level=1',
'buildtype=debugoptimized' ])
+=== release 1.19.90 ===
+
+2022-01-28 14:28:35 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * NEWS:
+ * RELEASE:
+ * gst-omx.doap:
+ * meson.build:
+ Release 1.19.90
+
+2022-01-28 14:28:28 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * ChangeLog:
+ Update ChangeLogs for 1.19.90
+
2022-01-05 02:07:59 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
* meson.build:
GStreamer 1.20 Release Notes
GStreamer 1.20 has not been released yet. It is scheduled for release in
-November 2021.
+late January / early February 2022.
1.19.x is the unstable development version that is being developed in
-the git main branch and which will eventually result in 1.20, and 1.19.3
-is the current development release in that series
-
-It is expected that feature freeze will be in early October 2021,
-followed by one or two 1.19.9x pre-releases and the new 1.20 stable
-release around October/November 2021.
+the git main branch and which will eventually result in 1.20, and
+1.19.90 is the first release candidate in that series (1.20rc1).
1.20 will be backwards-compatible to the stable 1.18, 1.16, 1.14, 1.12,
-1.10, 1.8, 1.6,, 1.4, 1.2 and 1.0 release series.
+1.10, 1.8, 1.6, 1.4, 1.2 and 1.0 release series.
See https://gstreamer.freedesktop.org/releases/1.20/ for the latest
version of this document.
-Last updated: Monday 1 November 2021, 01:00 UTC (log)
+Last updated: Wednesday 26 January 2022, 01:00 UTC (log)
Introduction
Highlights
-- this section will be completed in due course
+- Development in GitLab was switched to a single git repository
+ containing all the modules
+- GstPlay: new high-level playback library, replaces GstPlayer
+- WebM Alpha decoding support
+- Encoding profiles can now be tweaked with additional
+ application-specified element properties
+- Compositor: multi-threaded video conversion and mixing
+- RTP header extensions: unified support in RTP depayloader and
+ payloader base classes
+- SMPTE 2022-1 2-D Forward Error Correction support
+- Smart encoding (passthrough) support for VP8, VP9, H.265 in
+ encodebin and transcodebin
+- Runtime compatibility support for libsoup2 and libsoup3 (libsoup3
+ support experimental)
+- Video decoder subframe support
+- Video decoder automatic packet-loss, data corruption, and keyframe
+ request handling for RTP / WebRTC / RTSP
+- MP4 and Matroska muxers now support profile/level/resolution changes
+ for H264/H265 input streams (i.e. codec data changing on the fly)
+- MP4 muxing mode that initially creates a fragmented mp4 which is
+ converted to a regular mp4 on EOS
+- Audio support for the WebKit Port for Embedded (WPE) web page source
+ element
+- CUDA based video color space convert and rescale elements and
+ upload/download elements
+- NVIDIA memory:NVMM support for OpenGL glupload and gldownload
+ elements
+- Many WebRTC improvements
+- The new VA-API plugin implemention fleshed out with more decoders
+ and new postproc elements
+- AppSink API to retrieve events in addition to buffers and buffer
+ lists
+- AppSrc gained more configuration options for the internal queue
+ (leakiness, limits in buffers and time, getters to read current
+ levels)
+- Updated Rust bindings and many new Rust plugins
+- Improved support for custom minimal GStreamer builds
+- Support build against FFmpeg 5.0
+- Linux Stateless CODEC support gained MPEG2 and VP9
+- Windows Direct3D11/DXVA decoder gained AV1 and MPEG2 support
+- Lots of new plugins, features, performance improvements and bug
+ fixes
Major new features and changes
Noteworthy new features and API
-- this section will be filled in in due course
+- gst_element_get_request_pad() has been deprecated in favour of the
+ newly-added gst_element_request_pad_simple() which does the exact
+ same thing but has a less confusing name that hopefully makes clear
+ that the function request a new pad rather than just retrieves an
+ already-existing request pad.
+
+Development in GitLab was switched to a single git repository containing all the modules
+
+The GStreamer multimedia framework is a set of libraries and plugins
+split into a number of distinct modules which are released independently
+and which have so far been developed in separate git repositories in
+freedesktop.org GitLab.
+
+In addition to these separate git repositories there was a gst-build
+module that would use the Meson build systems’s subproject feature to
+download each individual module and then build everything in one go. It
+would also provide an uninstalled development environment that made it
+easy to work on GStreamer and use or test versions other than the
+system-installed GStreamer version.
+
+All of these modules have now (as of 28 September 2021) been merged into
+a single git repository (“Mono repository” or “monorepo”) which should
+simplify development workflows and continuous integration, especially
+where changes need to be made to multiple modules at once.
+
+This mono repository merge will primarily affect GStreamer developers
+and contributors and anyone who has workflows based on the GStreamer git
+repositories.
+
+The Rust bindings and Rust plugins modules have not been merged into the
+mono repository at this time because they follow a different release
+cycle.
+
+The mono repository lives in the existing GStreamer core git repository
+in GitLab in the new main branch and all future development will happen
+on this branch.
+
+Modules will continue to be released as separate tarballs.
+
+For more details, please see the GStreamer mono repository FAQ.
+
+GstPlay: new high-level playback library replacing GstPlayer
+
+- GstPlay is a new high-level playback library that replaces the older
+ GstPlayer API. It is basically the same API as GstPlayer but
+ refactored to use bus messages for application notifications instead
+ of GObject signals. There is still a signal adapter object for those
+ who prefer signals. Since the existing GstPlayer API is already in
+ use in various applications, it didn’t seem like a good idea to
+ break it entirely. Instead a new API was added, and it is expected
+ that this new GstPlay API will be moved to gst-plugins-base in
+ future.
+
+- The existing GstPlayer API is scheduled for deprecation and will be
+ removed at some point in the future (e.g. in GStreamer 1.24), so
+ application developers are urged to migrate to the new GstPlay API
+ at their earliest convenience.
+
+WebM alpha decoding
+
+- Implement WebM alpha decoding (VP8/VP9 with alpha), which required
+ support and additions in various places. This is supported both with
+ software decoders and hardware-accelerated decoders.
+
+- VP8/VP9 don’t support alpha components natively in the codec, so the
+ way this is implemented in WebM is by encoding the alpha plane with
+ transparency data as a separate VP8/VP9 stream. Inside the WebM
+ container (a variant of Matroska) this is coded as a single video
+ track with the “normal” VP8/VP9 video data making up the main video
+ data and each frame of video having an encoded alpha frame attached
+ to it as extra data ("BlockAdditional").
+
+- matroskademux has been extended extract this per-frame alpha side
+ data and attach it in form of a GstVideoCodecAlphaMeta to the
+ regular video buffers. Note that this new meta is specific to this
+ VP8/VP9 alpha support and can’t be used to just add alpha support to
+ other codecs that don’t support it. Lastly, matroskademux also
+ advertises the fact that the streams contain alpha in the caps.
+
+- The new codecalpha plugin contains various bits of infrastructure to
+ support autoplugging and debugging:
+
+ - codecalphademux splits out the alpha stream from the metas on
+ the regular VP8/VP9 buffers
+ - alphacombine takes two decoded raw video streams (one alpha, one
+ the regular video) and combines it into a video stream with
+ alpha
+ - vp8alphadecodebin + vp9alphadecodebin are wrapper bins that use
+ the regular vp8dec and vp9dec software decoders to decode
+ regular and alpha streams and combine them again. To decodebin
+ these look like regular decoders which ju
+ - The V4L2 CODEC plugin has stateless VP8/VP9 decoders that can
+ decode both alpha and non-alpha stream with a single decoder
+ instance
+
+- A new AV12 video format was added which is basically NV12 with an
+ alpha plane, which is more convenient for many hardware-accelerated
+ decoders.
+
+- Watch Nicolas Dufresne’s LCA 2022 talk “Bringing WebM Alpha support
+ to GStreamer” for all the details and a demo.
+
+RTP Header Extensions Base Class and Automatic Header Extension Handling in RTP Payloaders and Depayloaders
+
+- RTP Header Extensions are specified in RFC 5285 and provide a way to
+ add small pieces of data to RTP packets in between the RTP header
+ and the RTP payload. This is often used for per-frame metadata,
+ extended timestamps or other application-specific extra data. There
+ are several commonly-used extensions specified in various RFCs, but
+ senders are free to put any kind of data in there, as long as sender
+ and receiver both know what that data is. Receivers that don’t know
+ about the header extensions will just skip the extra data without
+ ever looking at it. These header extensions can often be combined
+ with any kind of payload format, so may need to be supported by many
+ RTP payloader and depayloader elements.
+
+- Inserting and extracting RTP header extension data has so far been a
+ bit inconvenient in GStreamer: There are functions to add and
+ retrieve RTP header extension data from RTP packets, but nothing
+ works automatically, even for common extensions. People would have
+ to do the insertion/extraction either in custom elements
+ before/after the RTP payloader/depayloader, or inside pad probes,
+ which isn’t very nice.
+
+- This release adds various pieces of new infrastructure for generic
+ RTP header extension handling, as well as some implementations for
+ common extensions:
+
+ - GstRTPHeaderExtension is a new helper base class for reading and
+ writing RTP header extensions. Nominally this subclasses
+ GstElement, but only so these extensions are stored in the
+ registry where they can be looked up by URI or name. They don’t
+ have pads and don’t get added to the pipeline graph as an
+ element.
+
+ - "add-extension" and "clear-extension" action signals on RTP
+ payloaders and depayloaders for manual extension management
+
+ - The "request-extension" signal will be emitted if an extension
+ is encountered that requires explicit mapping by the application
+
+ - new "auto-header-extension" property on RTP payloaders and
+ depayloaders for automatic handling of known header extensions.
+ This is enabled by default. The extensions must be signalled via
+ caps / SDP.
+
+ - RTP header extension implementations:
+
+ - rtphdrextclientaudiolevel: Client-to-Mixer Audio Level
+ Indication (RFC 6464) (also see below)
+ - rtphdrextcolorspace: Color Space extension, extends RTP
+ packets with color space and high dynamic range (HDR)
+ information
+ - rtphdrexttwcc: Transport Wide Congestion Control support
+
+- gst_rtp_buffer_remove_extension_data() is a new helper function to
+ remove an RTP header extension from an RTP buffer
+
+- The existing gst_rtp_buffer_set_extension_data() now also supports
+ shrinking the extension data in size
+
+AppSink and AppSrc improvements
+
+- appsink: new API to pull events out of appsink in addition to
+ buffers and buffer lists.
+
+ There was previously no way for users to receive incoming events
+ from appsink properly serialised with the data flow, even if they
+ are serialised events. The reason for that is that the only way to
+ intercept events was via a pad probe on the appsink sink pad, but
+ there is also internal queuing inside of appsink, so it’s difficult
+ to ascertain the right order of everything in all cases.
+
+ There is now a new "new-serialized-event" signal which will be
+ emitted when there’s a new event pending (just like the existing
+ "new-sample" signal). The "emit-signals" property must be set to
+ TRUE in order to activate this (but it’s also fine to just pull from
+ the application thread without using the signals).
+
+ gst_app_sink_pull_object() and gst_app_sink_try_pull_object() can be
+ used to pull out either an event or a new sample carrying a buffer
+ or buffer list, whatever is next in the queue.
+
+ EOS events will be filtered and will not be returned. EOS handling
+ can be done the ususal way, same as with _pull_sample().
+
+- appsrc: allow configuration of internal queue limits in time and
+ buffers and add leaky mode.
+
+ There is internal queuing inside appsrc so the application thread
+ can push data into the element which will then be picked up by the
+ source element’s streaming thread and pushed into the pipeline from
+ that streaming thread. This queue is unlimited by default and until
+ now it was only possible to set a maximum size limit in bytes. When
+ that byte limit is reached, the pushing thread (application thread)
+ would be blocked until more space becomes available.
+
+ A limit in bytes is not particularly useful for many use cases, so
+ now it is possible to also configure limits in time and buffers
+ using the new "max-time" and "max-buffers" properties. Of course
+ there are also matching new read-only"current-level-buffers" and
+ "current-level-time properties" properties to query the current fill
+ level of the internal queue in time and buffers.
+
+ And as if that wasn’t enough the internal queue can also be
+ configured as leaky using the new "leaky-type" property. That way
+ when the queue is full the application thread won’t be blocked when
+ it tries to push in more data, but instead either the new buffer
+ will be dropped or the oldest data in the queue will be dropped.
+
+Better string serialization of nested GstCaps and GstStructures
+
+- New string serialisation format for structs and caps that can handle
+ nested structs and caps properly by using brackets to delimit nested
+ items (e.g. some-struct, some-field=[nested-struct, nested=true]).
+ Unlike the default format the new variant can also support more than
+ one level of nesting. For backwards-compatibility reasons the old
+ format is still output by default when serialising caps and structs
+ using the existing API. The new functions gst_caps_serialize() and
+ gst_structure_serialize() can be used to output strings in the new
+ format.
+
+Convenience API for custom GstMetas
+
+- New convenience API to register and create custom GstMetas:
+ gst_meta_register_custom() and gst_buffer_add_custom_meta(). Such
+ custom meta is backed by a GstStructure and does not require that
+ users of the API expose their GstMeta implementation as public API
+ for other components to make use of it. In addition, it provides a
+ simpler interface by ignoring the impl vs. api distinction that the
+ regular API exposes. This new API is meant to be the meta
+ counterpart to custom events and messages, and to be more convenient
+ than the lower-level API when the absolute best performance isn’t a
+ requirement. The reason it’s less performant than a “proper” meta is
+ that a proper meta is just a C struct in the end whereas this goes
+ through the GstStructure API which has a bit more overhead, which
+ for most scenarios is negligible however. This new API is useful for
+ experimentation or proprietary metas, but also has some limitations:
+ it can only be used if there’s a single producer of these metas;
+ it’s not allowed to register the same custom meta multiple times or
+ from multiple places.
+
+Additional Element Properties on Encoding Profiles
+
+- GstEncodingProfile: The new "element-properties" and
+ gst_encoding_profile_set_element_properties() API allows
+ applications to set additional element properties on encoding
+ profiles to configure muxers and encoders. So far the encoding
+ profile template was the only place where this could be specified,
+ but often what applications want to do is take a ready-made encoding
+ profile shipped by GStreamer or the application and then tweak the
+ settings on top of that, which is now possible with this API. Since
+ applications can’t always know in advance what encoder element will
+ be used in the end, it’s even possible to specify properties on a
+ per-element basis.
+
+ Encoding Profiles are used in the encodebin, transcodebin and
+ camerabin elements and APIs to configure output formats (containers
+ and elementary streams).
+
+Audio Level Indication Meta for RFC 6464
+
+- New GstAudioLevelMeta containing Audio Level Indication as per RFC
+ 6464
+
+- The level element has been updated to add GstAudioLevelMeta on
+ buffers if the "audio-level-meta" property is set to TRUE. This can
+ then in turn be picked up by RTP payloaders to signal the audio
+ level to receivers through RTP header extensions (see above).
+
+- New Client-to-Mixer Audio Level Indication (RFC6464) RTP Header
+ Extension which should be automatically created and used by RTP
+ payloaders and depayloaders if their "auto-header-extension"
+ property is enabled and if the extension is part of the RTP caps.
+
+Automatic packet loss, data corruption and keyframe request handling for video decoders
+
+- The GstVideoDecoder base class has gained various new APIs to
+ automatically handle packet loss and data corruption better by
+ default, especially in RTP, RTSP and WebRTC streaming scenarios, and
+ to give subclasses more control about how they want to handle
+ missing data:
+
+ - Video decoder subclasses can mark output frames as corrupted via
+ the new GST_VIDEO_CODEC_FRAME_FLAG_CORRUPTED flag
+
+ - A new "discard-corrupted-frames" property allows applications to
+ configure decoders so that corrupted frames are directly
+ discarded instead of being forwarded inside the pipeline. This
+ is a replacement for the "output-corrupt" property of the FFmpeg
+ decoders.
+
+ - RTP depayloaders can now signal to decoders that data is missing
+ when sending GAP events for lost packets. GAP events can be sent
+ for various reason in a GStreamer pipeline. Often they are just
+ used to let downstream elements know that there isn’t a buffer
+ available at the moment, so downstream elements can move on
+ instead of waiting for one. They are also sent by RTP
+ depayloaders in the case that packets are missing, however, and
+ so far a decoder was not able to differentiate the two cases.
+ This has been remedied now: GAP events can be decorated with
+ gst_event_set_gap_flags() and GST_GAP_FLAG_MISSING_DATA to let
+ decoders now what happened, and decoders can then use that in
+ some cases to handle missing data better.
+
+ - The GstVideoDecoder::handle_missing_data vfunc was added to
+ inform subclasses about packet loss or missing data and let them
+ handle it in their own way if they like.
+
+ - gst_video_decoder_set_needs_sync_point() lets subclasses signal
+ that they need the stream to start with a sync point. If
+ enabled, the base class will discard all non-sync point frames
+ in the beginning and after a flush and does not pass them to the
+ subclass. Furthermore, if the first frame is not a sync point,
+ the base class will try and request a sync frame from upstream
+ by sending a force-key-unit event (see next items).
+
+ - New "automatic-request-sync-points" and
+ "automatic-request-sync-point-flags" properties to automatically
+ request sync points when needed, e.g. on packet loss or if the
+ first frame is not a keyframe. Applications may want to enable
+ this on decoders operating in e.g. RTP/WebRTC/RTSP receiver
+ pipelines.
+
+ - The new "min-force-key-unit-interval" property can be used to
+ ensure there’s a minimal interval between keyframe requests to
+ upstream (and/or the sender) and we’re not flooding the sender
+ with key unit requests.
+
+ - gst_video_decoder_request_sync_point() allows subclasses to
+ request a new sync point (e.g. if they choose to do their own
+ missing data handling). This will still honour the
+ "min-force-key-unit-interval" property if set.
+
+Improved support for custom minimal GStreamer builds
+
+- Element registration and registration of other plugin features
+ inside plugin init functions has been improved in order to
+ facilitate minimal custom GStreamer builds.
+
+- A number of new macros have been added to declare and create
+ per-element and per-pluginfeature register functions in all plugins,
+ and then call those from the per-plugin plugin_init functions:
+
+ - GST_ELEMENT_REGISTER_DEFINE,
+ GST_DEVICE_PROVIDER_REGISTER_DEFINE,
+ GST_DYNAMIC_TYPE_REGISTER_DEFINE, GST_TYPE_FIND_REGISTER_DEFINE
+ for the actual registration call with GStreamer
+ - GST_ELEMENT_REGISTER, GST_DEVICE_PROVIDER_REGISTER,
+ GST_DYNAMIC_TYPE_REGISTER, GST_PLUGIN_STATIC_REGISTER,
+ GST_TYPE_FIND_REGISTER to call the registration function defined
+ by the REGISTER_DEFINE macro
+ - GST_ELEMENT_REGISTER_DECLARE,
+ GST_DEVICE_PROVIDER_REGISTER_DECLARE,
+ GST_DYNAMIC_TYPE_REGISTER_DECLARE,
+ GST_TYPE_FIND_REGISTER_DECLARE to declare the registration
+ function defined by the REGISTER_DEFINE macro
+ - and various variants for advanced use cases.
+
+- This means that applications can call the per-element and
+ per-pluginfeature registration functions for only the elements they
+ need instead of registering plugins as a whole with all kinds of
+ elements that may not be required (e.g. encoder and decoder instead
+ of just decoder). In case of static linking all unused functions and
+ their dependencies would be removed in this case by the linker,
+ which helps minimise binary size for custom builds.
+
+- gst_init() will automatically call a gst_init_static_plugins()
+ function if one exists.
+
+- See the GStreamer static build documentation and Stéphane’s blog
+ post Generate a minimal GStreamer build, tailored to your needs for
+ more details.
New elements
-- this section will be filled in in due course
+- New aesdec and aesenc elements for AES encryption and decryption in
+ a custom format.
+
+- New encodebin2 element with dynamic/sometimes source pads in order
+ to support the option of doing the muxing outside of encodebin,
+ e.g. in combination with a splitmuxsink.
+
+- New fakeaudiosink and videocodectestsink elements for testing and
+ debugging (see below for more details)
+
+- rtpisacpay, rtpisacdepay: new RTP payloader and depayloader for iSAC
+ audio codec
+
+- rtpst2022-1-fecdec, rtpst2022-1-fecenc: new elements providing SMPTE
+ 2022-1 2-D Forward Error Correction. More details in Mathieu’s blog
+ post.
+
+- isac: new plugin wrapping the Internet Speech Audio Codec reference
+ encoder and decoder from the WebRTC project.
+
+- asio: plugin for Steinberg ASIO (Audio Streaming Input/Output) API
+
+- gssrc, gssink: add source and sink for Google Cloud Storage
+
+- onnx: new plugin to apply ONNX neural network models to video
+
+- openaptx: aptX and aptX-HD codecs using libopenaptx (v0.2.0)
+
+- qroverlay, debugqroverlay: new elements that allows overlaying data
+ on top of video in form of a QR code
+
+- cvtracker: new OpenCV-based tracker element
+
+- av1parse, vp9parse: new parsers for AV1 and VP9 video
+
+- va: work on the new VA-API plugin implementation for
+ hardware-accelerated video decoding and encoding has continued at
+ pace, with various new decoders and filters having joined the
+ initial vah264dec:
+
+ - vah265dec: VA-API H.265 decoder
+ - vavp8dec: VA-API VP8 decoder
+ - vavp9dec: VA-API VP9 decoder
+ - vaav1dec: VA-API AV1 decoder
+ - vampeg2dec: VA-API MPEG-2 decoder
+ - vadeinterlace: : VA-API deinterlace filter
+ - vapostproc: : VA-API postproc filter (color conversion,
+ resizing, cropping, color balance, video rotation, skin tone
+ enhancement, denoise, sharpen)
+
+ See Víctor’s blog post “GstVA in GStreamer 1.20” for more details
+ and what’s coming up next.
+
+- vaapiav1dec: new AV1 decoder element (in gstreamer-vaapi)
+
+- msdkav1dec: hardware-accelerated AV1 decoder using the Intel Media
+ SDK / oneVPL
+
+- nvcodec plugin for NVIDIA NVCODEC API for hardware-accelerated video
+ encoding and decoding:
+
+ - cudaconvert, cudascale: new CUDA based video color space convert
+ and rescale elements
+ - cudaupload, cudadownload: new helper elements for memory
+ transfer between CUDA and system memory spaces
+ - nvvp8sldec, nvvp9sldec: new GstCodecs-based VP8/VP9 decoders
+
+- Various new hardware-accelerated elements for Windows:
+
+ - d3d11screencapturesrc: new desktop capture element, including a
+ GstDeviceProvider implementation to enumerate/select target
+ monitors for capture.
+ - d3d11av1dec and d3d11mpeg2dec: AV1 and MPEG-2 decoders
+ - d3d11deinterlace: deinterlacing filter
+ - d3d11compositor: video composing element
+ - see Windows section below for more details
+
+- new Rust plugins:
+
+ - audiornnoise: Removes noise from an audio stream
+ - awstranscribeparse: Parses AWS audio transcripts into timed text
+ buffers
+ - ccdetect: Detects if valid closed captions are present in a
+ closed captions stream
+ - cea608tojson: Converts CEA-608 Closed Captions to a JSON
+ representation
+ - cmafmux: CMAF fragmented MP4 muxer
+ - dashmp4mux: DASH fragmented MP4 muxer
+ - isofmp4mux: ISO fragmented MP4 muxer
+ - ebur128level: EBU R128 Loudness Level Measurement
+ - ffv1dec: FFV1 video decoder
+ - gtk4paintablesink: GTK4 video sink, which provides a
+ GdkPaintable that can be rendered in various widgets
+ - hlssink3: HTTP Live Streaming sink
+ - hrtfrender: Head-Related Transfer Function (HRTF) renderer
+ - hsvdetector: HSV colorspace detector
+ - hsvfilter: HSV colorspace filter
+ - jsongstenc: Wraps buffers containing any valid top-level JSON
+ structures into higher level JSON objects, and outputs those as
+ ndjson
+ - jsongstparse: Parses ndjson as output by jsongstenc
+ - jsontovtt: converts JSON to WebVTT subtitles
+ - regex: Applies regular expression operations on text
+ - roundedcorners: Adds rounded corners to video
+ - spotifyaudiosrc: Spotify source
+ - textahead: Display upcoming text buffers ahead (e.g. for
+ Karaoke)
+ - transcriberbin: passthrough bin that transcribes raw audio to
+ closed captions using awstranscriber and puts the captions as
+ metas onto the video
+ - tttojson: Converts timed text to a JSON representation
+ - uriplaylistbin: Playlist source bin
+ - webpdec-rs: WebP image decoder with animation support
+
+- New plugin codecalpha with elements to assist with WebM Alpha
+ decoding
+
+ - codecalphademux: Split stream with GstVideoCodecAlphaMeta into
+ two streams
+ - alphacombine: Combine two raw video stream (I420 or NV12) as one
+ stream with alpha channel (A420 or AV12)
+ - vp8alphadecodebin: A bin to handle software decoding of VP8 with
+ alpha
+ - vp9alphadecodebin: A bin to handle software decoding of VP9 with
+ alpha
+
+- New hardware accelerated elements for Linux:
+
+ - v4l2slmpeg2dec: Support for Linux Stateless MPEG2 decoders
+ - v4l2slvp9dec: Support for Linux Stateless VP9 decoders
+ - v4l2slvp8alphadecodebin: Support HW accelerated VP8 with alpha
+ layer decoding
+ - v4l2slvp9alphadecodebin: Support HW accelerated VP9 with alpha
+ layer decoding
New element features and additions
-- this section will be filled in in due course
+- assrender: handle more font mime types; better interaction with
+ matroskademux for embedded fonts
+
+- audiobuffersplit: Add support for specifying output buffer size in
+ bytes (not just duration)
+
+- audiolatency: new "samplesperbuffer" property so users can configure
+ the number of samples per buffer. The default value is 240 samples
+ which is equivalent to 5ms latency with a sample rate of 48000,
+ which might be larger than actual buffer size of audio capture
+ device.
+
+- audiomixer, audiointerleave, GstAudioAggregator: now keep a count of
+ samples that are dropped or processed as statistic and can be made
+ to post QoS messages on the bus whenever samples are dropped by
+ setting the "qos-messages" property on input pads.
+
+- audiomixer, compositor: improved handling of new inputs added at
+ runtime. New API was added to the GstAggregator base class to allow
+ subclasses to opt into an aggregation mode where inactive pads are
+ ignored when processing input buffers
+ (gst_aggregator_set_ignore_inactive_pads(),
+ gst_aggregator_pad_is_inactive()). An “inactive pad” in this context
+ is a pad which, in live mode, hasn’t yet received a first buffer,
+ but has been waited on at least once. What would happen usually in
+ this case is that the aggregator would wait for data on this pad
+ every time, up to the maximum configured latency. This would
+ inadvertently push mixer elements in live mode to the configured
+ latency envelope and delay processing when new inputs are added at
+ runtime until these inputs have actually produced data. This is
+ usually undesirable. With this new API, new inputs can be added
+ (requested) and configured and they won’t delay the data processing.
+ Applications can opt into this new behaviour by setting the
+ "ignore-inactive-pads" property on compositor, audiomixer or other
+ GstAudioAggregator-based elements.
+
+- cccombiner: implement “scheduling” of captions. So far cccombiner’s
+ behaviour was essentially that of a funnel: it strictly looked at
+ input timestamps to associate together video and caption buffers.
+ Now it will try to smoothly schedule caption buffers in order to
+ have exactly one per output video buffer. This might involve
+ rewriting input captions, for example when the input is CDP then
+ sequence counters are rewritten, time codes are dropped and
+ potentially re-injected if the input video frame had a time code
+ meta. This can also lead to the input drifting from synchronization,
+ when there isn’t enough padding in the input stream to catch up. In
+ that case the element will start dropping old caption buffers once
+ the number of buffers in its internal queue reaches a certain limit
+ (configurable via the "max-scheduled" property). The new original
+ funnel-like behaviour can be restored by setting the "scheduling"
+ property to FALSE.
+
+- ccconverter: new "cdp-mode" property to specify which sections to
+ include in CDP packets (timecode, CC data, service info). Various
+ software, including ffmpeg’s Decklink support, fails parsing CDP
+ packets that contain anything but CC data in the CDP packets.
+
+- clocksync: new "sync-to-first" property for automatic timestamp
+ offset setup: if set clocksync will set up the "ts-offset" value
+ based on the first buffer and the pipeline’s running time when the
+ first buffer arrived. The newly configured "ts-offset" in this case
+ would be the value that allows outputting the first buffer without
+ waiting on the clock. This is useful for example to feed a non-live
+ input into an already-running pipeline.
+
+- compositor:
+
+ - multi-threaded input conversion and compositing. Set the
+ "max-threads" property to activate this.
+ - new "sizing-policy" property to support display aspect ratio
+ (DAR)-aware scaling. By default the image is scaled to fill the
+ configured destination rectangle without padding and without
+ keeping the aspect ratio. With sizing-policy=keep-aspect-ratio
+ the input image is scaled to fit the destination rectangle
+ specified by GstCompositorPad:{xpos, ypos, width, height}
+ properties preserving the aspect ratio. As a result, the image
+ will be centered in the destination rectangle with padding if
+ necessary.
+ - new "zero-size-is-unscaled" property on input pads. By default
+ pad width=0 or pad height=0 mean that the stream should not be
+ scaled in that dimension. But if the "zero-size-is-unscaled"
+ property is set to FALSE a width or height of 0 is instead
+ interpreted to mean that the input image on that pad should not
+ be composited, which is useful when creating animations where an
+ input image is made smaller and smaller until it disappears.
+ - improved handling of new inputs at runtime via
+ "ignore-inactive-pads"property (see above for details)
+ - allow output format with alpha even if none of the inputs have
+ alpha (also glvideomixer and other GstVideoAggregator
+ subclasses)
+
+- dashsink: add h265 codec support and signals for allowing custom
+ playlist/fragment output
+
+- decodebin3:
+
+ - improved decoder selection, especially for hardware decoders
+ - make input activation “atomic” when adding inputs dynamically
+ - better interleave handling: take into account decoder latency
+ for interleave size
+
+- decklink:
+
+ - Updated DeckLink SDK to 11.2 to support DeckLink 8K Pro
+ - decklinkvideosrc:
+ - More accurate and stable capture timestamps: use the
+ hardware reference clock time when the frame was finished
+ being captured instead of a clock time much further down the
+ road.
+ - Automatically detect widescreen vs. normal NTSC/PAL
+
+- encodebin:
+
+ - add “smart encoding” support for H.265, VP8 and VP9 (i.e. only
+ re-encode where needed and otherwise pass through encoded video
+ as-is).
+ - H264/H265 smart encoding improvements: respect user-specified
+ stream-format, but if not specified default to avc3/hvc1 with
+ in-band SPS/PPS/VPS signalling for more flexibility.
+ - new encodebin2 element with dynamic/sometimes source pads in
+ order to support the option of doing the muxing outside of
+ encodebin, e.g. in combination with splitmuxsink.
+ - add APIs to set element properties on encoding profiles (see
+ below)
+
+- errorignore: new "ignore-eos" property to also ignore FLOW_EOS from
+ downstream elements
+
+- giosrc: add support for growing source files: applications can
+ specify that the underlying file being read is growing by setting
+ the "is-growing" property. If set, the source won’t EOS when it
+ reaches the end of the file, but will instead start monitoring it
+ and will start reading data again whenever a change is detected. The
+ new "waiting-data" and "done-waiting-data" signals keep the
+ application informed about the current state.
+
+- gtksink, gtkglsink:
+
+ - scroll event support: forwarded as navigation events into the
+ pipeline
+ - "video-aspect-ratio-override" property to force a specific
+ aspect ratio
+ - "rotate-method" property and support automatic rotation based on
+ image tags
+
+- identity: new "stats" property allows applications to retrieve the
+ number of bytes and buffers that have passed through so far.
+
+- interlace: add support for more formats, esp 10-bit, 12-bit and
+ 16-bit ones
+
+- jack: new "low-latency" property for automatic latency-optimized
+ setting and "port-names" property to select ports explicitly
+
+- jpegdec: support output conversion to RGB using libjpeg-turbo (for
+ certain input files)
+
+- line21dec:
+
+ - "mode" property to control whether and how detected closed
+ captions should be inserted in the list of existing close
+ caption metas on the input frame (if any): add, drop, or
+ replace.
+ - "ntsc-only" property to only look for captions if video has NTSC
+ resolution
+
+- line21enc: new "remove-caption-meta" to remove metas from output
+ buffers after encoding the captions into the video data; support for
+ CDP closed captions
+
+- matroskademux, matroskamux: Add support for ffv1, a lossless
+ intra-frame video coding format.
+
+- matroskamux: accept in-band SPS/PPS/VPS for H264 and H265
+ (i.e. stream-format avc3 and hev1) which allows on-the-fly
+ profile/level/resolution changes.
+
+- matroskamux: new "cluster-timestamp-offset" property, useful for use
+ cases where the container timestamps should map to some absolute
+ wall clock time, for example.
+
+- rtpsrc: add "caps" property to allow explicit setting of the caps
+ where needed
+
+- mpegts: support SCTE-35 passthrough via new "send-scte35-events"
+ property on MPEG-TS demuxer tsdemux. When enabled, SCTE 35 sections
+ (eg ad placement opportunities) are forwarded as events donwstream
+ where they can be picked up again by mpegtsmux. This required a
+ semantic change in the SCTE-35 section API: timestamps are now in
+ running time instead of muxer pts.
+
+- tsdemux: Handle PCR-less MPEG-TS streams; more robust timestamp
+ handling in certain corner cases and for poorly muxed streams.
+
+- mpegtsmux:
+
+ - More conformance improvements to make MPEG-TS analyzers happy:
+ - PCR timing accuracy: Improvements to the way mpegtsmux
+ outputs PCR observations in CBR mode, so that a PCR
+ observation is always inserted when needed, so that we never
+ miss the configured pcr-interval, as that triggers various
+ MPEG-TS analyser errors.
+ - Improved PCR/SI scheduling
+ - Don’t write PCR until PAT/PMT are output to make sure streams
+ start cleanly with a PAT/PMT.
+ - Allow overriding the automatic PMT PID selection via
+ application-supplied PMT_%d fields in the prog-map
+ structure/property.
+
+- mp4mux:
+
+ - new "first-moov-then-finalise" mode for fragmented output where
+ the output will start with a self-contained moov atom for the
+ first fragment, and then produce regular fragments. Then at the
+ end when the file is finalised, the initial moov is invalidated
+ and a new moov is written covering the entire file. This way the
+ file is a “fragmented mp4” file while it is still being written
+ out, and remains playable at all times, but at the end it is
+ turned into a regular mp4 file (with former fragment headers
+ remaining as unused junk data in the file).
+ - support H.264 avc3 and H.265 hvc1 stream formats as input where
+ the codec data is signalled in-band inside the bitstream instead
+ of caps/file headers.
+ - support profile/level/resolution changes for H264/H265 input
+ streams (i.e. codec data changing on the fly). Each codec_data
+ is put into its own SampleTableEntry inside the stsd, unless the
+ input is in avc3 stream format in which case it’s written
+ in-band an not in the headers.
+
+- multifilesink: new ""min-keyframe-distance"" property to make
+ minimum distance between keyframes in next-file=key-frame mode
+ configurable instead of hard-coding it to 10 seconds.
+
+- mxfdemux has seen a big refactoring to support non-frame wrappings
+ and more accurate timestamp/seek handling for some formats
+
+- msdk plugin for hardware-accelerated video encoding and decoding
+ using the Intel Media SDK:
+
+ - oneVPL support (Intel oneAPI Video Processing Library)
+ - AV1 decoding support
+ - H.264 decoder now supports constrained-high and progressive-high
+ profiles
+ - H.264 encoder:
+ - more configuration options (properties):
+ "intra-refresh-type", "min-qp" , "max-qp", "p-pyramid",
+ "dblk-idc"
+ - H.265 encoder:
+ - can output main-still-picture profile
+ - now inserts HDR SEIs (mastering display colour volume and
+ content light level)
+ - more configuration options (properties):
+ "intra-refresh-type", "min-qp" , "max-qp", "p-pyramid",
+ "b-pyramid", "dblk-idc", "transform-skip"
+ - support for RGB 10bit format
+ - External bitrate control in encoders
+ - Video post proc element msdkvpp gained support for 12-bit pixel
+ formats P012_LE, Y212_LE and Y412_LE
+
+- nvh264sldec: interlaced stream support
+
+- openh264enc: support main, high, constrained-high and
+ progressive-high profiles
+
+- openjpeg: support for multithreaded decoding and encoding
+
+- rtspsrc: now supports IPv6 also for tunneled mode (RTSP-over-HTTP);
+ new "ignore-x-server-reply" property to ignore the
+ x-server-ip-address server header reply in case of HTTP tunneling,
+ as it is often broken.
+
+- souphttpsrc: Runtime compatibility support for libsoup2 and
+ libsoup3. libsoup3 is the latest major version of libsoup, but
+ libsoup2 and libsoup3 can’t co-exist in the same process because
+ there is no namespacing or versioning for GObject types. As a
+ result, it would be awkward if the GStreamer souphttpsrc plugin
+ linked to a specific version of libsoup, because it would only work
+ with applications that use the same version of libsoup. To make this
+ work, the soup plugin now tries to determine the libsoup version
+ used by the application (and its other dependencies) at runtime on
+ systems where GStreamer is linked dynamically. libsoup3 support is
+ still considered somewhat experimental at this point.
+
+- srtsrc, srtsink: add signals for the application to accept/reject
+ incoming connections
+
+- timeoverlay: new elapsed-running-time time mode which shows the
+ running time since the first running time (and each flush-stop).
+
+- udpsrc: new timestamping mode to retrieve packet receive timestamps
+ from the kernel via socket control messages (SO_TIMESTAMPNS) on
+ supported platforms
+
+- uritranscodebin: new setup-source and element-setup signals for
+ applications to configure elements used
+
+- v4l2codecs plugin gained support for 4x4 and 32x32 tile formats
+ enabling some platforms or direct renders. Important memory usage
+ improvement.
+
+- v4l2slh264dec now implements the final Linux uAPI as shipped on
+ Linux 5.11 and later.
+
+- valve: add "drop-mode" property and provide two new modes of
+ operation: in drop-mode=forward-sticky-events sticky events
+ (stream-start, segment, tags, caps, etc.) are forwarded downstream
+ even when dropping is enabled; drop-mode=transform-to-gap will in
+ addition also convert buffers into gap events when dropping is
+ enabled, which lets downstream elements know that time is advancing
+ and might allow for preroll in many scenarios. By default all events
+ and all buffers are dropped when dropping is enabled, which can
+ cause problems with caps negotiation not progressing or branches not
+ prerolling when dropping is enabled.
+
+- videocrop: support for many more pixel formats, e.g. planar YUV
+ formats with > 8bits and GBR* video formats; can now also accept
+ video not backed by system memory as long as downstream supports the
+ GstCropMeta
+
+- videotestsrc: new smpte-rp-219 pattern for SMPTE75 RP-219 conformant
+ color bars
+
+- vp8enc: finish support for temporal scalability: two new properties
+ ("temporal-scalability-layer-flags",
+ "temporal-scalability-layer-sync-flags") and a unit change on the
+ "temporal-scalability-target-bitrate" property (now expects bps);
+ also make temporal scalability details available to RTP payloaders
+ as buffer metadata.
+
+- vp9enc: new properties to tweak encoder performance:
+
+ - "aq-mode" to configure adaptive quantization modes
+ - "frame-parallel-decoding" to configure whether to create a
+ bitstream that reduces decoding dependencies between frames
+ which allows staged parallel processing of more than one video
+ frames in the decoder. (Defaults to TRUE)
+ - "row-mt", "tile-columns" and "tile-rows" so multithreading can
+ be enabled on a per-tile basis, instead of on a per tile-column
+ basis. In combination with the new "tile-rows" property, this
+ allows the encoder to make much better use of the available CPU
+ power.
+
+- vp9dec, vp9enc: add support for 10-bit 4:2:0 and 4:2:2 YUV, as well
+ as 8-bit 4:4:4
+
+- vp8enc, vp9enc now default to “good quality” for the deadline
+ property rather then “best quality”. Having the deadline set to best
+ quality causes the encoder to be absurdly slow, most real-life users
+ will prefer good-enough quality with better performance instead.
+
+- wpesrc:
+
+ - implement audio support: a new sometimes source pad will be
+ created for each audio stream created by the web engine.
+ - move wpesrc to wpevideosrc and add a wrapper bin wpesrc to also
+ support audio
+ - also handles web:// URIs now (same as cefsrc)
+ - post messages with the estimated load progress on the bus
-Plugin and library moves
+- x265enc: add negative DTS support, which means timestamps are now
+ offset by 1h same as with x264enc
+
+RTP Payloaders and Depayloaders
+
+- rtpisacpay, rtpisacdepay: new RTP payloader and depayloader for iSAC
+ audio codec
+
+- rtph264depay:
+
+ - new "request-keyframe" property to make the depayloader
+ automatically request a new keyframe from the sender on packet
+ loss, consistent with the new property on rtpvp8depay.
+ - new "wait-for-keyframe" property to make depayloader wait for a
+ new keyframe at the beginning and after packet loss (only
+ effective if the depayloader outputs AUs), consistent with the
+ existing property on rtpvp8depay.
+
+- rtpopuspay, rtpopusdepay: support libwebrtc-compatible multichannel
+ audio in addition to the previously supported multichannel audio
+ modes
+
+- rtpopuspay: add DTX (Discontinuous Transmission) support
+
+- rtpvp8depay: new "request-keyframe" property to make the depayloader
+ automatically request a new keyframe from the sender on packet loss.
+
+- rtpvp8pay: temporal scaling support
-- this section will be filled in in due course
+- rtpvp9depay: Improved SVC handling (aggregate all layers)
+
+RTP Infrastructure
+
+- rtpst2022-1-fecdec, rtpst2022-1-fecenc: new elements providing SMPTE
+ 2022-1 2-D Forward Error Correction. More details in Mathieu’s blog
+ post.
+
+- rtpreddec: BUNDLE support
+
+- rtpredenc, rtpulpfecenc: add support for Transport-wide Congestion
+ Control (TWCC)
+
+- rtpsession: new "twcc-feedback-interval" property to allow RTCP TWCC
+ reports to be scheduled on a timer instead of per marker-bit.
+
+Plugin and library moves
- There were no plugin moves or library moves in this cycle.
The following elements or plugins have been removed:
-- this section will be filled in in due course
-
- The ofa audio fingerprinting plugin has been removed. The MusicIP
database has been defunct for years so this plugin is likely neither
useful nor used by anyone.
Miscellaneous API additions
-- this section will be filled in in due course
+Core
+
+- gst_buffer_new_memdup() is a convenience function for the
+ widely-used gst_buffer_new_wrapped(g_memdup(data,size),size)
+ pattern.
+
+- gst_caps_features_new_single() creates a new single GstCapsFeatures,
+ avoiding the need to use the vararg function with NULL terminator
+ for simple cases.
+
+- gst_element_type_set_skip_documentation() can be used by plugins to
+ signal that certain elements should not be included in the GStreamer
+ plugin documentation. This is useful for plugins where elements are
+ registered dynamically based on hardware capabilities and/or where
+ the available plugins and properties vary from system to system.
+ This is used in the d3d11 plugin for example to ensure that only the
+ list of default elements is advertised in the documentation.
+
+- gst_type_find_suggest_empty_simple() is a new convenience function
+ for typefinders for cases where there’s only a media type and no
+ other fields.
+
+- New API to create elements and set properties at construction time,
+ which is not only convenient, but also allows GStreamer elements to
+ have construct-only properties: gst_element_factory_make_full(),
+ gst_element_factory_make_valist(),
+ gst_element_factory_make_with_properties(),
+ gst_element_factory_create_full(),
+ gst_element_factory_create_valist(),
+ gst_element_factory_create_with_properties().
+
+- GstSharedTaskPool: new “shared” task pool subclass with slightly
+ different default behaviour than the existing GstTaskPool which
+ would create unlimited number of threads for new tasks. The shared
+ taskpool creates up to N threads (default: 1) and then distributes
+ pending tasks to those threads round-robin style, and blocks if no
+ thread is available. It is possible to join tasks. This can be used
+ by plugins to implement simple multi-threaded processing and is used
+ for the new multi-threaded video conversion and compositing done in
+ GstVideoAggregator, videoconverter and compositor.
+
+Plugins Base Utils library
+
+- GstDiscoverer:
+
+ - gst_discoverer_container_info_get_tags() was added to retrieve
+ global/container tags (vs. per-stream tags). Per-Stream tags can
+ be retrieved via the existing
+ gst_discoverer_stream_info_get_tags().
+ gst_discoverer_info_get_tags(), which for many files returns a
+ confusing mix of stream and container tags, has been deprecated
+ in favour of the container/stream-specific functions.
+ - gst_discoverer_stream_info_get_stream_number() returns a unique
+ integer identifier for a given stream within the given
+ GstDiscoverer context. (If this matches the stream number inside
+ the container bitstream that’s by coincidence and not by
+ design.)
+
+- gst_pb_utils_get_caps_description_flags() can be used to query
+ whether certain caps represent a container, audio, video, image,
+ subtitles, tags, or something else. This only works for formats
+ known to GStreamer.
+
+- gst_pb_utils_get_file_extension_from_caps() returns a possible file
+ extension for given caps.
+
+- gst_codec_utils_h264_get_profile_flags_level(): Parses profile,
+ flags, and level from H264 AvcC codec_data. The format of H264 AVCC
+ extradata/sequence_header is documented in the ITU-T H.264
+ specification section 7.3.2.1.1 as well as in ISO/IEC 14496-15
+ section 5.3.3.1.2.
+
+- gst_codec_utils_caps_get_mime_codec() to convert caps to a RFC 6381
+ compatible MIME codec string codec. Useful for providing the codecs
+ field inside the Content-Type HTTP header for containerized formats,
+ such as mp4 or matroska.
+
+GStreamer OpenGL integration library and plugins
+
+- glcolorconvert: added suppport for converting the video formats
+ A420, AV12, BGR, BGRA, RGBP and BGRP.
+
+- Added support to GstGLBuffer for persistent buffer mappings where a
+ Pixel Buffer Object (PBO) can be mapped by both the CPU and the GPU.
+ This removes a memcpy() when uploading textures or vertices
+ particularly when software decoders (e.g. libav) are direct
+ rendering into our memory. Improves transfer performance
+ significantly. Requires OpenGL 4.4, GL_ARB_buffer_storage or
+ GL_EXT_buffer_storage
+
+- Added various helper functions for handling 4x4 matrices of affine
+ transformations as used by GstVideoAffineTransformationMeta.
+
+- Add support to GstGLContext for allowing the application to control
+ the config (EGLConfig, GLXConfig, etc) used when creating the OpenGL
+ context. This allows the ability to choose between RGB16 or RGB10A2
+ or RGBA8 back/front buffer configurations that were previously
+ hardcoded. GstGLContext also supports retrieving the configuration
+ it was created with or from an externally provide OpenGL context
+ handle. This infrastructure is also used to create a compatible
+ config from an application/externally provided OpenGL context in
+ order to improve compatibility with other OpenGL frameworks and GUI
+ toolkits. A new environment variable GST_GL_CONFIG was also added to
+ be able to request a specific configuration from the command line.
+ Note: different platforms will have different functionality
+ available.
+
+- Add support for choosing between EGL and WGL at runtime when running
+ on Windows. Previously this was a build-time switch. Allows use in
+ e.g. Gtk applications on Windows that target EGL/ANGLE without
+ recompiling GStreamer. gst_gl_display_new_with_type() can be used by
+ applications to choose a specific display type to use.
+
+- Build fixes to explicitly check for Broadcom-specific libraries on
+ older versions of the Raspberry Pi platform. The Broadcom OpenGL ES
+ and EGL libraries have different filenames. Using the vc4 Mesa
+ driver on the Raspberry Pi is not affected.
+
+- Added support to glupload and gldownload for transferring RGBA
+ buffers using the memory:NVMM available on the Nvidia Tegra family
+ of embedded devices.
+
+- Added support for choosing libOpenGL and libGLX as used in a GLVND
+ environment on unix-based platforms. This allows using desktop
+ OpenGL and EGL without pulling in any GLX symbols as would be
+ required with libGL.
+
+Video library
+
+- New raw video formats:
+
+ - AV12 (NV12 with alpha plane)
+ - RGBP and BGRP (planar RGB formats)
+ - ARGB64 variants with specified endianness instead of host
+ endianness:
+ - ARGB64_LE, ARGB64_BE
+ - RGBA64_BE, RGBA64_LE
+ - BGRA64_BE, BGRA64_LE
+ - ABGR64_BE, ABGR64_LE
+
+- gst_video_orientation_from_tag() is new convenience API to parse the
+ image orientation from a GstTagList.
+
+- GstVideoDecoder subframe support (see below)
+
+- GstVideoCodecState now also carries some HDR metadata
+
+- Ancillary video data: implement transform functions for AFD/Bar
+ metas, so they will be forwarded in more cases
+
+MPEG-TS library
+
+This library only handles section parsing and such, see above for
+changes to the actual mpegtsmux and mpegtsdemux elements.
+
+- many additions and improvements to SCTE-35 section parsing
+- new API for fetching extended descriptors:
+ gst_mpegts_find_descriptor_with_extension()
+- add support for SIT sections (Selection Information Tables)
+- expose event-from-section constructor gst_event_new_mpegts_section()
+- parse Audio Preselection Descriptor needed for Dolby AC-4
+
+GstWebRTC library + webrtcbin
+
+- Change the way in which sink pads and transceivers are matched
+ together to support easier usage. If a pad is created without a
+ specific index (i.e. using sink_%u as the pad template), then an
+ available compatible transceiver will be searched for. If a specific
+ index is requested (i.e. sink_1) then if a transceiver for that
+ m-line already exists, that transceiver must match the new sink pad
+ request. If there is no transceiver available in either scenario, a
+ new transceiver is created. If a mixture of both sink_1 and sink_%u
+ requests result in an impossible situation, an error will be
+ produced at pad request time or from create offer/answer.
+
+- webrtcbin now uses regular ICE nomination instead of libnice’s
+ default of aggressive ICE nomination. Regular ICE nomination is the
+ default recommended by various relevant standards and improves
+ connectivity in specific network scenarios.
+
+- Add support for limiting the port range used for RTP with the
+ addition of the min-rtp-port and max-rtp-port properties on the ICE
+ object.
+
+- Expose the SCTP transport as a property on webrtcbin to more closely
+ match the WebRTC specification.
+
+- Added support for taking into account the data channel transport
+ state when determining the value of the "connection-state" property.
+ Previous versions of the WebRTC spec did not include the data
+ channel state when computing this value.
+
+- Add configuration for choosing the size of the underlying sockets
+ used for transporting media data
+
+- Always advertise support for the transport-cc RTCP feedback protocol
+ as rtpbin supports it. For full support, the configured caps (input
+ or through codec-preferences) need to include the relevant RTP
+ header extension.
+
+- Numerous fixes to caps and media handling to fail-fast when an
+ incompatible situation is detected.
+
+- Improved support for attaching the required media after a remote
+ offer has been set.
+
+- Add support for dynamically changing the amount of FEC used for a
+ particular stream.
+
+- webrtcbin now stops further SDP processing at the first error it
+ encounters.
+
+- Completed support for either local or the remote closing a data
+ channel.
+
+- Various fixes when performing BUNDLEing of the media streams in
+ relation to RTX and FEC usage.
+
+- Add support for writing out QoS DSCP marking on outgoing packets to
+ improve reliability in some network scenarios.
+
+- Improvements to the statistics returned by the get-stats signal
+ including the addition of the raw statistics from the internal
+ RTPSource, the TWCC stats when available.
+
+- The webrtc library does not expose any objects anymore with public
+ fields. Instead properties have been added to replace that
+ functionality. If you are accessing such fields in your application,
+ switch to the corresponding properties.
+
+GstCodecs and Video Parsers
+
+- Support for render delays to improve throughput across all CODECs
+ (used with NVDEC and V4L2).
+- lots of improvements to parsers and the codec parsing decoder base
+ classes (H264, H265, VP8, VP9, AV1, MPEG-2) used for various
+ hardware-accelerate decoder APIs.
+
+Bindings support
+
+- gst_allocation_params_new() allocates a GstAllocationParams struct
+ on the heap. This should only be used by bindings (and freed via
+ gst_allocation_params_free() then). In C code you would allocate
+ this on the stack and only init it in place.
+
+- gst_debug_log_literal() can be used to log a string to the debug log
+ without going through any printf format expansion and associated
+ overhead. This is mostly useful for bindings such as the Rust
+ bindings which may have done their own formatting already .
+
+- Provide non-inlined versions of refcounting APIs for various
+ GStreamer mini objects, so that they can be consumed by bindings
+ (e.g. gstreamer-sharp): gst_buffer_ref, gst_buffer_unref,
+ gst_clear_buffer, gst_buffer_copy, gst_buffer_replace,
+ gst_buffer_list_ref, gst_buffer_list_unref, gst_clear_buffer_list,
+ gst_buffer_list_copy, gst_buffer_list_replace, gst_buffer_list_take,
+ gst_caps_ref, gst_caps_unref, gst_clear_caps, gst_caps_replace,
+ gst_caps_take, gst_context_ref, gst_context_unref, gst_context_copy,
+ gst_context_replace, gst_event_replace, gst_event_steal,
+ gst_event_take, gst_event_ref, gst_event_unref, gst_clear_event,
+ gst_event_copy, gst_memory_ref, gst_memory_unref, gst_message_ref,
+ gst_message_unref, gst_clear_message, gst_message_copy,
+ gst_message_replace, gst_message_take, gst_promise_ref,
+ gst_promise_unref, gst_query_ref, gst_query_unref, gst_clear_query,
+ gst_query_copy, gst_query_replace, gst_query_take, gst_sample_ref,
+ gst_sample_unref, gst_sample_copy, gst_tag_list_ref,
+ gst_tag_list_unref, gst_clear_tag_list, gst_tag_list_replace,
+ gst_tag_list_take, gst_uri_copy, gst_uri_ref, gst_uri_unref,
+ gst_clear_uri.
+
+- expose a GType for GstMiniObject
+
+- gst_device_provider_probe() now returns non-floating device object
+
+API Deprecations
+
+- gst_element_get_request_pad() has been deprecated in favour of the
+ newly-added gst_element_request_pad_simple() which does the exact
+ same thing but has a less confusing name that hopefully makes clear
+ that the function request a new pad rather than just retrieves an
+ already-existing request pad.
+
+- gst_discoverer_info_get_tags(), which for many files returns a
+ confusing mix of stream and container tags, has been deprecated in
+ favour of the container-specific and stream-specific functions,
+ gst_discoverer_container_info_get_tags() and
+ gst_discoverer_stream_info_get_tags().
+
+- gst_video_sink_center_rect() was deprecated in favour of the more
+ generic newly-added gst_video_center_rect().
+
+- The GST_MEMORY_FLAG_NO_SHARE flag has been deprecated, as it tends
+ to cause problems and prevents sub-buffering. If pooling or lifetime
+ tracking is required, memories should be allocated through a custom
+ GstAllocator instead of relying on the lifetime of the buffers the
+ memories were originally attached to, which is fragile anyway.
+
+- The GstPlayer high-level playback library is being replaced with the
+ new GstPlay library (see above). GstPlayer should be considered
+ deprecated at this point and will be marked as such in the next
+ development cycle. Applications should be ported to GstPlay.
+
+- Gstreamer Editing Services: ges_video_transition_set_border(),
+ ges_video_transition_get_border()
+ ges_video_transition_set_inverted()
+ ges_video_transition_is_inverted() have been deprecated, use
+ ges_timeline_element_set_children_properties() instead.
Miscellaneous performance, latency and memory optimisations
-- this section will be filled in in due course
+More video conversion fast paths
-Miscellaneous other changes and enhancements
+- v210 ↔ I420, YV12, Y42B, UYVY and YUY2
+- A420 → RGB
+
+Less jitter when waiting on the system clock
+
+- Better system clock wait accuracy, less jitter: where available,
+ clock_nanosleep is used for higher accuracy for waits below 500
+ usecs, and waits below 2ms will first use the regular waiting system
+ and then clock_nanosleep for the remainder. The various wait
+ implementation have a latency ranging from 50 to 500+ microseconds.
+ While this is not a major issue when dealing with a low number of
+ waits per second (for ex: video), it does introduce a non-negligible
+ jitter for synchronisation of higher packet rate systems.
+
+Video decoder subframe support
-- this section will be filled in in due course
+- The GstVideoDecoder base class gained API to process input at the
+ sub-frame level. That way video decoders can start decoding slices
+ before they have received the full input frame in its entirety (to
+ the extent this is supported by the codec, of course). This helps
+ with CPU utilisation and reduces latency.
-Tracing framework and debugging improvements
+- This functionality is now being used in the OpenJPEG JPEG 2000
+ decoder, the FFmpeg H.264 decoder (in case of NAL-aligned input) and
+ the OpenMAX H.264/H.265 decoders (in case of NAL-aligned input).
-- this section will be filled in in due course
+Miscellaneous other changes and enhancements
+
+- GstDeviceMonitor no longer fails to start just because one of the
+ device providers failed to start. That could happen for example on
+ systems where the pulseaudio device provider is installed, but
+ pulseaudio isn’t actually running but ALSA is used for audio
+ instead. In the same vein the device monitor now keeps track of
+ which providers have been started (via the new
+ gst_device_provider_is_started()) and only stops actually running
+ device providers when stopping the device monitor.
+
+- On embedded systems it can be useful to create a registry that can
+ be shared and read by multiple processes running as different users.
+ It is now possible to set the new GST_REGISTRY_MODE environment
+ variable to specify the file mode for the registry file, which by
+ default is set to be only user readable/writable.
+
+- GstNetClientClock will signal lost sync in case the remote time
+ resets (e.g. because device power cycles), by emitting the “synced”
+ signal with synced=FALSE parameter, so applications can take action.
+
+- gst_value_deserialize_with_pspec() allows deserialization with a
+ hint for what the target GType should be. This allows for example
+ passing arrays of flags through the command line or
+ gst_util_set_object_arg(), eg: foo="<bar,bar+baz>".
+
+- It’s now allowed to create an empty GstVideoOverlayComposition
+ without any rectangles by passing a NULL rectangle to
+ gst_video_overlay_composition_new(). This is useful for bindings and
+ simplifies application code in some places.
+
+Tracing framework, debugging and testing improvements
+
+- New factories tracer to list loaded elements (and other plugin
+ features). This can be useful to collect a list of elements needed
+ for an application, which then in turn can be used to create a
+ tailored minimal GStreamer build that contains just the elements
+ needed and nothing else.
+- New plugin-feature-loaded tracing hook for use by tracers like the
+ new factories tracer
+
+- GstHarness: Add gst_harness_set_live() so that harnesses can be set
+ to non-live and return is-live=false in latency queries if needed.
+ Default behaviour is to always return is-live=true in latency
+ queries.
+
+- navseek: new "hold-eos" property. When enabled, the element will
+ hold back an EOS event until the next keystroke (via navigation
+ events). This can be used to keep a video sink showing the last
+ frame of a video pipeline until a key is pressed instead of tearing
+ it down immediately on EOS.
+
+- New fakeaudiosink element: mimics an audio sink and can be used for
+ testing and CI pipelines on systems where no audio system is
+ installed or running. It differs from fakesink in that it only
+ support audio caps and syncs to the clock by default like a normal
+ audio sink. It also implements the GstStreamVolume interface like
+ most audio sinks do.
+
+- New videocodectestsink element for video codec conformance testing:
+ Calculates MD5 checksums for video frames and skips any padding
+ whilst doing so. Can optionally also write back the video data with
+ padding removed into a file for easy byte-by-byte comparison with
+ reference data.
Tools
-- this section will be filled in in due course
+gst-inspect-1.0
+
+- Can sort the list of plugins by passing --sort=name as command line
+ option
+
+gst-launch-1.0
+
+- will now error out on top-level properties that don’t exist and
+ which were silently ignored before
+- On Windows the high-resolution clock is enabled now, which provides
+ better clock and timer performance on Windows (see Windows section
+ below for more details).
+
+gst-play-1.0
+
+- New --start-position command line argument to start playback from
+ the specified position
+- Audio can be muted/unmuted in interactive mode by pressing the m
+ key.
+- On Windows the high-resolution clock is enabled now (see Windows
+ section below for more details)
+
+gst-device-monitor-1.0
+
+- New --include-hidden command line argument to also show “hidden”
+ device providers
+
+ges-launch-1.0
+
+- New interactive mode that allows seeking and such. Can be disabled
+ by passing the --no-interactive argument on the command line.
+- Option to forward tags
+- Allow using an existing clip to determine the rendering format (both
+ topology and profile) via new --profile-from command line argument.
GStreamer RTSP server
-- this section will be filled in in due course
+- GstRTSPMediaFactory gained API to disable RTCP
+ (gst_rtsp_media_factory_set_enable_rtcp(), "enable-rtcp" property).
+ Previously RTCP was always allowed for all RTSP medias. With this
+ change it is possible to disable RTCP completely, no matter if the
+ client wants to do RTCP or not.
+
+- Make a mount point of / work correctly. While not allowed by the
+ RTSP 2 spec, the RTSP 1 spec is silent on this and it is used in the
+ wild. It is now possible to use / as a mount path in
+ gst-rtsp-server, e.g. rtsp://example.com/ would work with this now.
+ Note that query/fragment parts of the URI are not necessarily
+ correctly handled, and behaviour will differ between various
+ client/server implementations; so use it if you must but don’t bug
+ us if it doesn’t work with third party clients as you’d hoped.
+
+- multithreading fixes (races, refcounting issues, deadlocks)
+
+- ONVIF audio backchannel fixes
+
+- ONVIF trick mode optimisations
+
+- rtspclientsink: new "update-sdp" signal that allows updating the SDP
+ before sending it to the server via ANNOUNCE. This can be used to
+ add additional metadata to the SDP, for example. The order and
+ number of medias must not be changed, however.
GStreamer VAAPI
-- this section will be filled in in due course
+- new AV1 decoder element (vaapiav1dec)
+
+- H264 decoder: handle stereoscopic 3D video with frame packing
+ arrangement SEI messages
+
+- H265 encoder: added Screen Content Coding extensions support
+
+- H265 decoder: gained MAIN_444_12 profile support (decoded to
+ Y412_LE), and 4:2:2 12-bits support (decoded to Y212_LE)
+
+- vaapipostproc: gained BT2020 color standard support
+
+- vaapidecode: now generates caps templates dynamically at runtime in
+ order to advertise actually supported caps instead of all
+ theoretically supported caps.
+
+- GST_VAAPI_DRM_DEVICE environment variable to force a specified DRM
+ device when a DRM display is used. It is ignored when other types of
+ displays are used. By default /dev/dri/renderD128 is used for DRM
+ display.
GStreamer OMX
-- this section will be filled in in due course
+- subframe support in H.264/H.265 decoders
GStreamer Editing Services and NLE
-- this section will be filled in in due course
+- framepositioner: new "operator" property to access blending modes in
+ the compositor
+- timeline: Implement snapping to markers
+- smart-mixer: Add support for d3d11compositor and glvideomixer
+- titleclip: add "draw-shadow" child property
+- ges:// URI support to define a timeline from a description.
+- command-line-formatter
+ - Add track management to timeline description
+ - Add keyframe support
+- ges-launch-1.0:
+ - Add an interactive mode where we can seek etc…
+ - Add option to forward tags
+ - Allow using an existing clip to determine the rendering format
+ (both topology and profile) via new --profile-from command line
+ argument.
+- Fix static build
GStreamer validate
-- this section will be filled in in due course
+- report: Add a way to force backtraces on reports even if not a
+ critical issue (GST_VALIDATE_ISSUE_FLAGS_FORCE_BACKTRACE)
+- Add a flag to gst_validate_replace_variables_in_string() allow
+ defining how to resolve variables in structs
+- Add gst_validate_bin_monitor_get_scenario() to get the bin monitor
+ scenario, which is useful for applications that use Validate
+ directly.
+- Add an expected-values parameter to wait, message-type=XX allowing
+ more precise filtering of the message we are waiting for.
+- Add config file support: each test can now use a config file for the
+ given media file used to test.
+- Add support to check properties of object properties
+- scenario: Add an "action-done" signal to signal when an action is
+ done
+- scenario: Add a "run-command" action type
+- scenario: Allow forcing running action on idle from scenario file
+- scenario: Allow iterating over arrays in foreach
+- scenario: Rename ‘interlaced’ action to ‘non-blocking’
+- scenario: Add a non-blocking flag to the wait signal
GStreamer Python Bindings
-- this section will be filled in in due course
+- Fixes for Python 3.10
+- Various build fixes
+- at least one known breaking change caused by g-i annotation changes
+ (see below)
GStreamer C# Bindings
-- this section will be filled in in due course
+- Fix GstDebugGraphDetails enum
+- Updated to latests GtkSharp
+- Updated to include GStreamer 1.20 API
GStreamer Rust Bindings and Rust Plugins
-The GStreamer Rust bindings are released separately with a different
-release cadence that’s tied to gtk-rs, but the latest release has
-already been updated for the upcoming new GStreamer 1.20 API.
-
-gst-plugins-rs, the module containing GStreamer plugins written in Rust,
-has also seen lots of activity with many new elements and plugins.
-
-What follows is a list of elements and plugins available in
-gst-plugins-rs, so people don’t miss out on all those potentially useful
-elements that have no C equivalent.
-
-- FIXME: add new elements
-
-Rust audio plugins
-
-- audiornnoise: New element for audio denoising which implements the
- noise removal algorithm of the Xiph RNNoise library, in Rust
-- rsaudioecho: Port of the audioecho element from gst-plugins-good
- rsaudioloudnorm: Live audio loudness normalization element based on
- the FFmpeg af_loudnorm filter
-- claxondec: FLAC lossless audio codec decoder element based on the
- pure-Rust claxon implementation
-- csoundfilter: Audio filter that can use any filter defined via the
- Csound audio programming language
-- lewtondec: Vorbis audio decoder element based on the pure-Rust
- lewton implementation
-
-Rust video plugins
-
-- cdgdec/cdgparse: Decoder and parser for the CD+G video codec based
- on a pure-Rust CD+G implementation, used for example by karaoke CDs
-- cea608overlay: CEA-608 Closed Captions overlay element
-- cea608tott: CEA-608 Closed Captions to timed-text (e.g. VTT or SRT
- subtitles) converter
-- tttocea608: CEA-608 Closed Captions from timed-text converter
-- mccenc/mccparse: MacCaption Closed Caption format encoder and parser
-- sccenc/sccparse: Scenarist Closed Caption format encoder and parser
-- dav1dec: AV1 video decoder based on the dav1d decoder implementation
- by the VLC project
-- rav1enc: AV1 video encoder based on the fast and pure-Rust rav1e
- encoder implementation
-- rsflvdemux: Alternative to the flvdemux FLV demuxer element from
- gst-plugins-good, not feature-equivalent yet
-- rsgifenc/rspngenc: GIF/PNG encoder elements based on the pure-Rust
- implementations by the image-rs project
-
-Rust text plugins
-
-- textwrap: Element for line-wrapping timed text (e.g. subtitles) for
- better screen-fitting, including hyphenation support for some
- languages
-
-Rust network plugins
-
-- reqwesthttpsrc: HTTP(S) source element based on the Rust
- reqwest/hyper HTTP implementations and almost feature-equivalent
- with the main GStreamer HTTP source souphttpsrc
-- s3src/s3sink: Source/sink element for the Amazon S3 cloud storage
-- awstranscriber: Live audio to timed text transcription element using
- the Amazon AWS Transcribe API
-
-Generic Rust plugins
-
-- sodiumencrypter/sodiumdecrypter: Encryption/decryption element based
- on libsodium/NaCl
-- togglerecord: Recording element that allows to pause/resume
- recordings easily and considers keyframe boundaries
-- fallbackswitch/fallbacksrc: Elements for handling potentially
- failing (network) sources, restarting them on errors/timeout and
- showing a fallback stream instead
-- threadshare: Set of elements that provide alternatives for various
- existing GStreamer elements but allow to share the streaming threads
- between each other to reduce the number of threads
-- rsfilesrc/rsfilesink: File source/sink elements as replacements for
- the existing filesrc/filesink elements
+- The GStreamer Rust bindings are released separately with a different
+ release cadence that’s tied to gtk-rs, but the latest release has
+ already been updated for the upcoming new GStreamer 1.20 API (v1_20
+ feature).
+
+- gst-plugins-rs, the module containing GStreamer plugins written in
+ Rust, has also seen lots of activity with many new elements and
+ plugins. See the New Elements section above for a list of new Rust
+ elements.
Build and Dependencies
- Meson 0.59 or newer is required to build GStreamer now.
-- FIXME: this section will be filled in in due course
+- The GLib requirement has been bumped to GLib 2.56 or newer (from
+ March 2018).
+
+- The wpe plugin now requires wpe >= 2.28 and wpebackend-fdo >= 1.8
Explicit opt-in required for build of certain plugins with (A)GPL dependencies
See Building plugins with (A)GPL-licensed dependencies for more details
and a non-exhaustive list of plugins affected.
-gst-build: replaced by Monorepo
+gst-build: replaced by mono repository
-- this section will be filled in in due course
-
-- FIXME: describe + link to Monorepo FAQ
+See mono repository section above and the GStreamer mono repository FAQ.
Cerbero
on platforms where dependencies are not readily available, such as
Windows, Android, iOS and macOS.
-General improvements
+General Cerbero improvements
-- this section will be filled in in due course
+- Plugin removed: libvisual
+- New plugins: rtpmanagerbad and rist
-macOS / iOS
+macOS / iOS specific Cerbero improvements
-- this section will be filled in in due course
+- XCode 12 support
+- macOS OS release support is now future-proof, similar to iOS
+- macOS Apple Silicon (ARM64) cross-compile support has been added
+- macOS Apple Silicon (ARM64) native support is currently experimental
-Windows
+Windows specific Cerbero improvements
-- this section will be filled in in due course
+- Visual Studio 2022 support has been added
+- bootstrap is faster since it requires building fewer build-tools
+ recipes on Windows
+- package is faster due to better scheduling of recipe stages and
+ elimination of unnecessary autotools regeneration
+- The following plugins are no longer built on Windows:
+ - a52dec (another decoder is still available in libav)
+ - dvdread
+ - resindvd
Windows MSI installer
-- this section will be filled in in due course
+- no major changes
-Linux
+Linux specific Cerbero improvements
-- this section will be filled in in due course
+- Fedora, Debian OS release support is now more future-proof
+- Amazon Linux 2 support has been added
-Android
+Android specific Cerbero improvements
-- this section will be filled in in due course
+- no major changes
Platform-specific changes and improvements
Android
-- this section will be filled in in due course
+- No major changes
macOS and iOS
-- this section will be filled in in due course
+- applemedia: add ProRes support to vtenc and vtdec
Windows
-- this section will be filled in in due course
+- On Windows the high-resolution clock is enabled now in the
+ gst-launch-1.0 and gst-play-1.0 command line tools, which provides
+ better clock and timer performance on Windows, at the cost of higher
+ power consumption. By default, without the high-resolution clock
+ enabled, the timer precision on Windows is system-dependent and may
+ be as bad as 15ms which is not good enough for many multimedia
+ applications. Developers may want to do the same in their Windows
+ applications if they think it’s a good idea for their application
+ use case, and depending on the Windows version they target. This is
+ not done automatically by GStreamer because on older Windows
+ versions (pre-Windows 10) this affects a global Windows setting and
+ also there’s a power consumption vs. performance trade-off that may
+ differ from application to application.
+
+- dxgiscreencapsrc now supports resolution changes
+
+- The wasapi2 audio plugin was rewritten and now has a higher rank
+ than the old wasapi plugin since it has a number of additional
+ features such as automatic stream routing, and no
+ known-but-hard-to-fix issues. The plugin is always built if the
+ Windows 10 SDK is available now.
+
+- The wasapi device providers now detect and notify dynamic device
+ additions/removals
+
+- d3d11screencapturesrc: new desktop capture element, including
+ GstDeviceProvider implementation to enumerate/select target monitors
+ for capture.
+
+- Direct3D11/DXVA decoder now supports AV1 and MPEG2 codecs
+ (d3d11av1dec, d3d11mpeg2dec)
+
+- VP9 decoding got more reliable and stable thanks to a newly written
+ codec parser
+
+- Support for decoding interlaced H.264/AVC streams
+
+- Hardware-accelerated video deinterlacing (d3d11deinterlace) and
+ video mixing (d3d11compositor)
+
+- Video mixing with the Direct3D11 API (d3d11compositor)
+
+- MediaFoundation API based hardware encoders gained the ability to
+ receive Direct3D11 textures as an input
+
+- Seungha’s blog post “GStreamer ❤ Windows: A primer on the cool stuff
+ you’ll find in the 1.20 release” describes many of the
+ Windows-related improvements in more detail
Linux
-- this section will be filled in in due course
+- bluez: LDAC Bluetooth audio codec support in a2dpsink and avdtpsink,
+ as well as an LDAC RTP payloader (rtpldacpay) and an LDAC audio
+ encoder (ldacenc)
+
+- kmssink: gained support for NV24, NV61, RGB16/BGR16 formats;
+ auto-detect NVIDIA Tegra driver
Documentation improvements
-- this section will be filled in in due course
+- hardware-accelerated GPU plugins will now no longer always list all
+ the element variants for all available GPUs, since those are
+ system-dependent and it’s confusing for users to see those in the
+ documentation just because the GStreamer developer who generated the
+ docs had multiple GPUs to play with at the time. Instead just show
+ the default elements.
-Possibly Breaking Changes
+Possibly Breaking and Other Noteworthy Behavioural Changes
+
+- gst_parse_launch(), gst_parse_bin_from_description() and friends
+ will now error out when setting properties that don’t exist on
+ top-level bins. They were silently ignored before.
+
+- The GstWebRTC library does not expose any objects anymore with
+ public fields. Instead properties have been added to replace that
+ functionality. If you are accessing such fields in your application,
+ switch to the corresponding properties.
-- FIXME: this section will be filled in in due course
-- MPEG-TS SCTE-35 API changes (FIXME: flesh out)
-- gst_parse_launch() and friends now error out on non-existing
- properties on top-level bins where they would silently fail and
- ignore those before.
- playbin and uridecodebin now emit the source-setup signal before the
element is added to the bin and linked so that the source element is
already configured before any scheduling query comes in, which is
- useful for elements such as appsrc or giostreamsrc. (Merge Request)
+ useful for elements such as appsrc or giostreamsrc.
+
+- The source element inside urisourcebin (used inside uridecodebin3
+ which is used inside playbin3) is no longer called "source". This
+ shouldn’t affect anyone hopefully, because there’s a "setup-source"
+ signal to configure the source element and no one should rely on
+ names of internal elements anyway.
+
+- The vp8enc element now expects bps (bits per second) for the
+ "temporal-scalability-target-bitrate" property, which is consistent
+ with the "target-bitrate" property. Since additional configuration
+ is required with modern libvpx to make temporal scaling work anyway,
+ chances are that very few people will have been using this property
+
+- vp8enc and vp9enc now default to “good quality” for the "deadline"
+ property rather then “best quality”. Having the deadline set to best
+ quality causes the encoder to be absurdly slow, most real-life users
+ will want the good quality tradeoff instead.
+
+- The experimental GstTranscoder library API in gst-plugins-bad was
+ changed from a GObject signal-based notification mechanism to a
+ GstBus/message-based mechanism akin to GstPlayer/GstPlay.
+
+- MPEG-TS SCTE-35 API: semantic change for SCTE-35 splice commands:
+ timestamps passed by the application should be in running time now,
+ since users of the API can’t really be expected to predict the local
+ PTS of the muxer.
+
+- The GstContext used by souphttpsrc to share the session between
+ multiple element instances has changed. Previously it provided
+ direct access to the internal SoupSession object, now it only
+ provides access to an opaque, internal type. This change is
+ necessary because SoupSession is not thread-safe at all and can’t be
+ shared safely between arbitrary external code and souphttpsrc.
+
+- Python bindings: GObject-introspection related Annotation fixes have
+ led to a case of a GstVideo.VideoInfo-related function signature
+ changing in the Python bindings (possibly one or two other cases
+ too). This is for a function that should never have been exposed in
+ the first place though, so the bindings are being updated to throw
+ an exception in that case, and the correct replacement API has been
+ added in form of an override.
Known Issues
-- this section will be filled in in due course
-
-- There are a couple of known WebRTC-related regressions/blockers:
-
- - webrtc: DTLS setup with Chrome is broken
- - webrtcbin: First keyframe is usually lost
+- nothing in particular at this point (but also see possibly breaking
+ changes section above)
Contributors
-- this section will be filled in in due course
+Aaron Boxer, Adam Leppky, Adam Williamson, Alba Mendez, Alejandro
+González, Aleksandr Slobodeniuk, Alexander Vandenbulcke, Alex Ashley,
+Alicia Boya García, Andika Triwidada, Andoni Morales Alastruey, Andrew
+Wesie, Andrey Moiseev, Antonio Ospite, Antonio Rojas, Arthur Crippa
+Búrigo, Arun Raghavan, Ashley Brighthope, Axel Kellermann, Baek, Bastien
+Nocera, Bastien Reboulet, Benjamin Gaignard, Bing Song, Binh Truong,
+Biswapriyo Nath, Brad Hards, Brad Smith, Brady J. Garvin, Branko
+Subasic, Camilo Celis Guzman, Chris Bass, ChrisDuncanAnyvision, Chris
+White, Corentin Damman, Daniel Almeida, Daniel Knobe, Daniel Stone,
+david, David Fernandez, David Keijser, David Phung, Devarsh Thakkar,
+Dinesh Manajipet, Dmitry Samoylov, Dmitry Shusharin, Dominique Martinet,
+Doug Nazar, Ederson de Souza, Edward Hervey, Emmanuel Gil Peyrot,
+Enrique Ocaña González, Ezequiel Garcia, Fabian Orccon, Fabrice
+Fontaine, Fernando Jimenez Moreno, Florian Karydes, Francisco Javier
+Velázquez-García, François Laignel, Frederich Munch, Fredrik Pålsson,
+George Kiagiadakis, Georg Lippitsch, Göran Jönsson, Guido Günther,
+Guillaume Desmottes, Guiqin Zou, Haakon Sporsheim, Haelwenn (lanodan)
+Monnier, Haihao Xiang, Haihua Hu, Havard Graff, He Junyan, Helmut
+Januschka, Henry Wilkes, Hosang Lee, Hou Qi, Ignacio Casal Quinteiro,
+Igor Kovalenko, Ilya Kreymer, Imanol Fernandez, Jacek Tomaszewski, Jade
+Macho, Jakub Adam, Jakub Janků, Jan Alexander Steffens (heftig), Jan
+Schmidt, Jason Carrete, Jason Pereira, Jay Douglass, Jeongki Kim, Jérôme
+Laheurte, Jimmi Holst Christensen, Johan Sternerup, John Hassell, John
+Lindgren, John-Mark Bell, Jonathan Matthew, Jordan Petridis, Jose
+Quaresma, Julian Bouzas, Julien, Kai Uwe Broulik, Kasper Steensig
+Jensen, Kellermann Axel, Kevin Song, Khem Raj, Knut Inge Hvidsten, Knut
+Saastad, Kristofer Björkström, Lars Lundqvist, Lawrence Troup, Lim Siew
+Hoon, Lucas Stach, Ludvig Rappe, Luis Paulo Fernandes de Barros, Luke
+Yelavich, Mads Buvik Sandvei, Marc Leeman, Marco Felsch, Marek Vasut,
+Marian Cichy, Marijn Suijten, Marius Vlad, Markus Ebner, Mart Raudsepp,
+Matej Knopp, Mathieu Duponchelle, Matthew Waters, Matthieu De Beule,
+Mengkejiergeli Ba, Michael de Gans, Michael Olbrich, Michael Tretter,
+Michal Dzik, Miguel Paris, Mikhail Fludkov, mkba, Nazar Mokrynskyi,
+Nicholas Jackson, Nicola Murino, Nicolas Dufresne, Niklas Hambüchen,
+Nikolay Sivov, Nirbheek Chauhan, Olivier Blin, Olivier Crete, Olivier
+Crête, Paul Goulpié, Per Förlin, Peter Boba, P H, Philippe Normand,
+Philipp Zabel, Pieter Willem Jordaan, Piotrek Brzeziński, Rafał
+Dzięgiel, Rafostar, raghavendra, Raghavendra, Raju Babannavar, Raleigh
+Littles III, Randy Li, Randy Li (ayaka), Ratchanan Srirattanamet, Raul
+Tambre, reed.lawrence, Ricky Tang, Robert Rosengren, Robert Swain, Robin
+Burchell, Roman Sivriver, R S Nikhil Krishna, Ruben Gonzalez, Ruslan
+Khamidullin, Sanchayan Maity, Scott Moreau, Sebastian Dröge, Sergei
+Kovalev, Seungha Yang, Sid Sethupathi, sohwan.park, Sonny Piers, Staz M,
+Stefan Brüns, Stéphane Cerveau, Stephan Hesse, Stian Selnes, Stirling
+Westrup, Théo MAILLART, Thibault Saunier, Tim, Timo Wischer, Tim-Philipp
+Müller, Tim Schneider, Tobias Ronge, Tom Schoonjans, Tulio Beloqui,
+tyler-aicradle, U. Artie Eoff, Ung, Val Doroshchuk, VaL Doroshchuk,
+Víctor Manuel Jáquez Leal, Vivek R, Vivia Nikolaidou, Vivienne
+Watermeier, Vladimir Menshakov, Will Miller, Wim Taymans, Xabier
+Rodriguez Calvar, Xavier Claessens, Xℹ Ruoyao, Yacine Bandou, Yinhang
+Liu, youngh.lee, youngsoo.lee, yychao, Zebediah Figura, Zhang yuankun,
+Zhang Yuankun, Zhao, Zhao Zhili, , Aleksandar Topic, Antonio Ospite,
+Bastien Nocera, Benjamin Gaignard, Brad Hards, Carlos Falgueras García,
+Célestin Marot, Corentin Damman, Corentin Noël, Daniel Almeida, Daniel
+Knobe, Danny Smith, Dave Piché, Dmitry Osipenko, Fabrice Fontaine,
+fjmax, Florian Zwoch, Guillaume Desmottes, Haihua Hu, Heinrich Kruger,
+He Junyan, Jakub Adam, James Cowgill, Jan Alexander Steffens (heftig),
+Jean Felder, Jeongki Kim, Jiri Uncovsky, Joe Todd, Jordan Petridis,
+Krystian Wojtas, Marc-André Lureau, Marcin Kolny, Marc Leeman, Mark
+Nauwelaerts, Martin Reboredo, Mathieu Duponchelle, Matthew Waters,
+Mengkejiergeli Ba, Michael Gruner, Nicolas Dufresne, Nirbheek Chauhan,
+Olivier Crête, Philippe Normand, Rafał Dzięgiel, Ralf Sippl, Robert
+Mader, Sanchayan Maity, Sangchul Lee, Sebastian Dröge, Seungha Yang,
+Stéphane Cerveau, Teh Yule Kim, Thibault Saunier, Thomas Klausner, Timo
+Wischer, Tim-Philipp Müller, Tobias Reineke, Tomasz Andrzejak, Trung Do,
+Tyler Compton, Ung, Víctor Manuel Jáquez Leal, Vivia Nikolaidou, Wim
+Taymans, wngecn, Wonchul Lee, wuchang li, Xavier Claessens, Xi Ruoyao,
+Yoshiharu Hirose, Zhao,
… and many others who have contributed bug reports, translations, sent
suggestions or helped testing.
1.20.0
-1.20.0 is scheduled to be released around October/November 2021.
+1.20.0 is scheduled to be released around early February 2022.
Schedule for 1.22
unstable development version leading up to the stable 1.22 release. The
development of 1.21/1.22 will happen in the git main branch.
-The plan for the 1.22 development cycle is yet to be confirmed.
+The plan for the 1.22 development cycle is yet to be confirmed. Assuming
+no major project-wide reorganisations in the 1.22 cycle we might try and
+aim for a release around August 2022.
1.22 will be backwards-compatible to the stable 1.20, 1.18, 1.16, 1.14,
1.12, 1.10, 1.8, 1.6, 1.4, 1.2 and 1.0 release series.
------------------------------------------------------------------------
These release notes have been prepared by Tim-Philipp Müller with
-contributions from …
+contributions from Matthew Waters, Nicolas Dufresne, Nirbheek Chauhan,
+Sebastian Dröge and Seungha Yang.
License: CC BY-SA 4.0
-This is GStreamer gst-omx 1.19.3.
+This is GStreamer gst-omx 1.19.90.
GStreamer 1.19 is the development branch leading up to the next major
stable version which will be 1.20.
<release>
<Version>
+ <revision>1.19.90</revision>
+ <branch>main</branch>
+ <name></name>
+ <created>2022-01-28</created>
+ <file-release rdf:resource="https://gstreamer.freedesktop.org/src/gst-omx/gst-omx-1.19.90.tar.xz" />
+ </Version>
+ </release>
+
+ <release>
+ <Version>
<revision>1.19.3</revision>
<branch>main</branch>
<name></name>
project('gst-omx', 'c',
- version : '1.19.3.1',
+ version : '1.19.90',
meson_version : '>= 0.59',
default_options : [ 'warning_level=1',
'buildtype=debugoptimized' ])
+=== release 1.19.90 ===
+
+2022-01-28 14:28:35 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * NEWS:
+ * RELEASE:
+ * gst-plugins-bad.doap:
+ * meson.build:
+ Release 1.19.90
+
+2022-01-28 14:28:28 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * ChangeLog:
+ Update ChangeLogs for 1.19.90
+
2022-01-27 11:22:54 +0100 Víctor Manuel Jáquez Leal <vjaquez@igalia.com>
* sys/va/gstvavpp.c:
GStreamer 1.20 Release Notes
GStreamer 1.20 has not been released yet. It is scheduled for release in
-November 2021.
+late January / early February 2022.
1.19.x is the unstable development version that is being developed in
-the git main branch and which will eventually result in 1.20, and 1.19.3
-is the current development release in that series
-
-It is expected that feature freeze will be in early October 2021,
-followed by one or two 1.19.9x pre-releases and the new 1.20 stable
-release around October/November 2021.
+the git main branch and which will eventually result in 1.20, and
+1.19.90 is the first release candidate in that series (1.20rc1).
1.20 will be backwards-compatible to the stable 1.18, 1.16, 1.14, 1.12,
-1.10, 1.8, 1.6,, 1.4, 1.2 and 1.0 release series.
+1.10, 1.8, 1.6, 1.4, 1.2 and 1.0 release series.
See https://gstreamer.freedesktop.org/releases/1.20/ for the latest
version of this document.
-Last updated: Monday 1 November 2021, 01:00 UTC (log)
+Last updated: Wednesday 26 January 2022, 01:00 UTC (log)
Introduction
Highlights
-- this section will be completed in due course
+- Development in GitLab was switched to a single git repository
+ containing all the modules
+- GstPlay: new high-level playback library, replaces GstPlayer
+- WebM Alpha decoding support
+- Encoding profiles can now be tweaked with additional
+ application-specified element properties
+- Compositor: multi-threaded video conversion and mixing
+- RTP header extensions: unified support in RTP depayloader and
+ payloader base classes
+- SMPTE 2022-1 2-D Forward Error Correction support
+- Smart encoding (passthrough) support for VP8, VP9, H.265 in
+ encodebin and transcodebin
+- Runtime compatibility support for libsoup2 and libsoup3 (libsoup3
+ support experimental)
+- Video decoder subframe support
+- Video decoder automatic packet-loss, data corruption, and keyframe
+ request handling for RTP / WebRTC / RTSP
+- MP4 and Matroska muxers now support profile/level/resolution changes
+ for H264/H265 input streams (i.e. codec data changing on the fly)
+- MP4 muxing mode that initially creates a fragmented mp4 which is
+ converted to a regular mp4 on EOS
+- Audio support for the WebKit Port for Embedded (WPE) web page source
+ element
+- CUDA based video color space convert and rescale elements and
+ upload/download elements
+- NVIDIA memory:NVMM support for OpenGL glupload and gldownload
+ elements
+- Many WebRTC improvements
+- The new VA-API plugin implemention fleshed out with more decoders
+ and new postproc elements
+- AppSink API to retrieve events in addition to buffers and buffer
+ lists
+- AppSrc gained more configuration options for the internal queue
+ (leakiness, limits in buffers and time, getters to read current
+ levels)
+- Updated Rust bindings and many new Rust plugins
+- Improved support for custom minimal GStreamer builds
+- Support build against FFmpeg 5.0
+- Linux Stateless CODEC support gained MPEG2 and VP9
+- Windows Direct3D11/DXVA decoder gained AV1 and MPEG2 support
+- Lots of new plugins, features, performance improvements and bug
+ fixes
Major new features and changes
Noteworthy new features and API
-- this section will be filled in in due course
+- gst_element_get_request_pad() has been deprecated in favour of the
+ newly-added gst_element_request_pad_simple() which does the exact
+ same thing but has a less confusing name that hopefully makes clear
+ that the function request a new pad rather than just retrieves an
+ already-existing request pad.
+
+Development in GitLab was switched to a single git repository containing all the modules
+
+The GStreamer multimedia framework is a set of libraries and plugins
+split into a number of distinct modules which are released independently
+and which have so far been developed in separate git repositories in
+freedesktop.org GitLab.
+
+In addition to these separate git repositories there was a gst-build
+module that would use the Meson build systems’s subproject feature to
+download each individual module and then build everything in one go. It
+would also provide an uninstalled development environment that made it
+easy to work on GStreamer and use or test versions other than the
+system-installed GStreamer version.
+
+All of these modules have now (as of 28 September 2021) been merged into
+a single git repository (“Mono repository” or “monorepo”) which should
+simplify development workflows and continuous integration, especially
+where changes need to be made to multiple modules at once.
+
+This mono repository merge will primarily affect GStreamer developers
+and contributors and anyone who has workflows based on the GStreamer git
+repositories.
+
+The Rust bindings and Rust plugins modules have not been merged into the
+mono repository at this time because they follow a different release
+cycle.
+
+The mono repository lives in the existing GStreamer core git repository
+in GitLab in the new main branch and all future development will happen
+on this branch.
+
+Modules will continue to be released as separate tarballs.
+
+For more details, please see the GStreamer mono repository FAQ.
+
+GstPlay: new high-level playback library replacing GstPlayer
+
+- GstPlay is a new high-level playback library that replaces the older
+ GstPlayer API. It is basically the same API as GstPlayer but
+ refactored to use bus messages for application notifications instead
+ of GObject signals. There is still a signal adapter object for those
+ who prefer signals. Since the existing GstPlayer API is already in
+ use in various applications, it didn’t seem like a good idea to
+ break it entirely. Instead a new API was added, and it is expected
+ that this new GstPlay API will be moved to gst-plugins-base in
+ future.
+
+- The existing GstPlayer API is scheduled for deprecation and will be
+ removed at some point in the future (e.g. in GStreamer 1.24), so
+ application developers are urged to migrate to the new GstPlay API
+ at their earliest convenience.
+
+WebM alpha decoding
+
+- Implement WebM alpha decoding (VP8/VP9 with alpha), which required
+ support and additions in various places. This is supported both with
+ software decoders and hardware-accelerated decoders.
+
+- VP8/VP9 don’t support alpha components natively in the codec, so the
+ way this is implemented in WebM is by encoding the alpha plane with
+ transparency data as a separate VP8/VP9 stream. Inside the WebM
+ container (a variant of Matroska) this is coded as a single video
+ track with the “normal” VP8/VP9 video data making up the main video
+ data and each frame of video having an encoded alpha frame attached
+ to it as extra data ("BlockAdditional").
+
+- matroskademux has been extended extract this per-frame alpha side
+ data and attach it in form of a GstVideoCodecAlphaMeta to the
+ regular video buffers. Note that this new meta is specific to this
+ VP8/VP9 alpha support and can’t be used to just add alpha support to
+ other codecs that don’t support it. Lastly, matroskademux also
+ advertises the fact that the streams contain alpha in the caps.
+
+- The new codecalpha plugin contains various bits of infrastructure to
+ support autoplugging and debugging:
+
+ - codecalphademux splits out the alpha stream from the metas on
+ the regular VP8/VP9 buffers
+ - alphacombine takes two decoded raw video streams (one alpha, one
+ the regular video) and combines it into a video stream with
+ alpha
+ - vp8alphadecodebin + vp9alphadecodebin are wrapper bins that use
+ the regular vp8dec and vp9dec software decoders to decode
+ regular and alpha streams and combine them again. To decodebin
+ these look like regular decoders which ju
+ - The V4L2 CODEC plugin has stateless VP8/VP9 decoders that can
+ decode both alpha and non-alpha stream with a single decoder
+ instance
+
+- A new AV12 video format was added which is basically NV12 with an
+ alpha plane, which is more convenient for many hardware-accelerated
+ decoders.
+
+- Watch Nicolas Dufresne’s LCA 2022 talk “Bringing WebM Alpha support
+ to GStreamer” for all the details and a demo.
+
+RTP Header Extensions Base Class and Automatic Header Extension Handling in RTP Payloaders and Depayloaders
+
+- RTP Header Extensions are specified in RFC 5285 and provide a way to
+ add small pieces of data to RTP packets in between the RTP header
+ and the RTP payload. This is often used for per-frame metadata,
+ extended timestamps or other application-specific extra data. There
+ are several commonly-used extensions specified in various RFCs, but
+ senders are free to put any kind of data in there, as long as sender
+ and receiver both know what that data is. Receivers that don’t know
+ about the header extensions will just skip the extra data without
+ ever looking at it. These header extensions can often be combined
+ with any kind of payload format, so may need to be supported by many
+ RTP payloader and depayloader elements.
+
+- Inserting and extracting RTP header extension data has so far been a
+ bit inconvenient in GStreamer: There are functions to add and
+ retrieve RTP header extension data from RTP packets, but nothing
+ works automatically, even for common extensions. People would have
+ to do the insertion/extraction either in custom elements
+ before/after the RTP payloader/depayloader, or inside pad probes,
+ which isn’t very nice.
+
+- This release adds various pieces of new infrastructure for generic
+ RTP header extension handling, as well as some implementations for
+ common extensions:
+
+ - GstRTPHeaderExtension is a new helper base class for reading and
+ writing RTP header extensions. Nominally this subclasses
+ GstElement, but only so these extensions are stored in the
+ registry where they can be looked up by URI or name. They don’t
+ have pads and don’t get added to the pipeline graph as an
+ element.
+
+ - "add-extension" and "clear-extension" action signals on RTP
+ payloaders and depayloaders for manual extension management
+
+ - The "request-extension" signal will be emitted if an extension
+ is encountered that requires explicit mapping by the application
+
+ - new "auto-header-extension" property on RTP payloaders and
+ depayloaders for automatic handling of known header extensions.
+ This is enabled by default. The extensions must be signalled via
+ caps / SDP.
+
+ - RTP header extension implementations:
+
+ - rtphdrextclientaudiolevel: Client-to-Mixer Audio Level
+ Indication (RFC 6464) (also see below)
+ - rtphdrextcolorspace: Color Space extension, extends RTP
+ packets with color space and high dynamic range (HDR)
+ information
+ - rtphdrexttwcc: Transport Wide Congestion Control support
+
+- gst_rtp_buffer_remove_extension_data() is a new helper function to
+ remove an RTP header extension from an RTP buffer
+
+- The existing gst_rtp_buffer_set_extension_data() now also supports
+ shrinking the extension data in size
+
+AppSink and AppSrc improvements
+
+- appsink: new API to pull events out of appsink in addition to
+ buffers and buffer lists.
+
+ There was previously no way for users to receive incoming events
+ from appsink properly serialised with the data flow, even if they
+ are serialised events. The reason for that is that the only way to
+ intercept events was via a pad probe on the appsink sink pad, but
+ there is also internal queuing inside of appsink, so it’s difficult
+ to ascertain the right order of everything in all cases.
+
+ There is now a new "new-serialized-event" signal which will be
+ emitted when there’s a new event pending (just like the existing
+ "new-sample" signal). The "emit-signals" property must be set to
+ TRUE in order to activate this (but it’s also fine to just pull from
+ the application thread without using the signals).
+
+ gst_app_sink_pull_object() and gst_app_sink_try_pull_object() can be
+ used to pull out either an event or a new sample carrying a buffer
+ or buffer list, whatever is next in the queue.
+
+ EOS events will be filtered and will not be returned. EOS handling
+ can be done the ususal way, same as with _pull_sample().
+
+- appsrc: allow configuration of internal queue limits in time and
+ buffers and add leaky mode.
+
+ There is internal queuing inside appsrc so the application thread
+ can push data into the element which will then be picked up by the
+ source element’s streaming thread and pushed into the pipeline from
+ that streaming thread. This queue is unlimited by default and until
+ now it was only possible to set a maximum size limit in bytes. When
+ that byte limit is reached, the pushing thread (application thread)
+ would be blocked until more space becomes available.
+
+ A limit in bytes is not particularly useful for many use cases, so
+ now it is possible to also configure limits in time and buffers
+ using the new "max-time" and "max-buffers" properties. Of course
+ there are also matching new read-only"current-level-buffers" and
+ "current-level-time properties" properties to query the current fill
+ level of the internal queue in time and buffers.
+
+ And as if that wasn’t enough the internal queue can also be
+ configured as leaky using the new "leaky-type" property. That way
+ when the queue is full the application thread won’t be blocked when
+ it tries to push in more data, but instead either the new buffer
+ will be dropped or the oldest data in the queue will be dropped.
+
+Better string serialization of nested GstCaps and GstStructures
+
+- New string serialisation format for structs and caps that can handle
+ nested structs and caps properly by using brackets to delimit nested
+ items (e.g. some-struct, some-field=[nested-struct, nested=true]).
+ Unlike the default format the new variant can also support more than
+ one level of nesting. For backwards-compatibility reasons the old
+ format is still output by default when serialising caps and structs
+ using the existing API. The new functions gst_caps_serialize() and
+ gst_structure_serialize() can be used to output strings in the new
+ format.
+
+Convenience API for custom GstMetas
+
+- New convenience API to register and create custom GstMetas:
+ gst_meta_register_custom() and gst_buffer_add_custom_meta(). Such
+ custom meta is backed by a GstStructure and does not require that
+ users of the API expose their GstMeta implementation as public API
+ for other components to make use of it. In addition, it provides a
+ simpler interface by ignoring the impl vs. api distinction that the
+ regular API exposes. This new API is meant to be the meta
+ counterpart to custom events and messages, and to be more convenient
+ than the lower-level API when the absolute best performance isn’t a
+ requirement. The reason it’s less performant than a “proper” meta is
+ that a proper meta is just a C struct in the end whereas this goes
+ through the GstStructure API which has a bit more overhead, which
+ for most scenarios is negligible however. This new API is useful for
+ experimentation or proprietary metas, but also has some limitations:
+ it can only be used if there’s a single producer of these metas;
+ it’s not allowed to register the same custom meta multiple times or
+ from multiple places.
+
+Additional Element Properties on Encoding Profiles
+
+- GstEncodingProfile: The new "element-properties" and
+ gst_encoding_profile_set_element_properties() API allows
+ applications to set additional element properties on encoding
+ profiles to configure muxers and encoders. So far the encoding
+ profile template was the only place where this could be specified,
+ but often what applications want to do is take a ready-made encoding
+ profile shipped by GStreamer or the application and then tweak the
+ settings on top of that, which is now possible with this API. Since
+ applications can’t always know in advance what encoder element will
+ be used in the end, it’s even possible to specify properties on a
+ per-element basis.
+
+ Encoding Profiles are used in the encodebin, transcodebin and
+ camerabin elements and APIs to configure output formats (containers
+ and elementary streams).
+
+Audio Level Indication Meta for RFC 6464
+
+- New GstAudioLevelMeta containing Audio Level Indication as per RFC
+ 6464
+
+- The level element has been updated to add GstAudioLevelMeta on
+ buffers if the "audio-level-meta" property is set to TRUE. This can
+ then in turn be picked up by RTP payloaders to signal the audio
+ level to receivers through RTP header extensions (see above).
+
+- New Client-to-Mixer Audio Level Indication (RFC6464) RTP Header
+ Extension which should be automatically created and used by RTP
+ payloaders and depayloaders if their "auto-header-extension"
+ property is enabled and if the extension is part of the RTP caps.
+
+Automatic packet loss, data corruption and keyframe request handling for video decoders
+
+- The GstVideoDecoder base class has gained various new APIs to
+ automatically handle packet loss and data corruption better by
+ default, especially in RTP, RTSP and WebRTC streaming scenarios, and
+ to give subclasses more control about how they want to handle
+ missing data:
+
+ - Video decoder subclasses can mark output frames as corrupted via
+ the new GST_VIDEO_CODEC_FRAME_FLAG_CORRUPTED flag
+
+ - A new "discard-corrupted-frames" property allows applications to
+ configure decoders so that corrupted frames are directly
+ discarded instead of being forwarded inside the pipeline. This
+ is a replacement for the "output-corrupt" property of the FFmpeg
+ decoders.
+
+ - RTP depayloaders can now signal to decoders that data is missing
+ when sending GAP events for lost packets. GAP events can be sent
+ for various reason in a GStreamer pipeline. Often they are just
+ used to let downstream elements know that there isn’t a buffer
+ available at the moment, so downstream elements can move on
+ instead of waiting for one. They are also sent by RTP
+ depayloaders in the case that packets are missing, however, and
+ so far a decoder was not able to differentiate the two cases.
+ This has been remedied now: GAP events can be decorated with
+ gst_event_set_gap_flags() and GST_GAP_FLAG_MISSING_DATA to let
+ decoders now what happened, and decoders can then use that in
+ some cases to handle missing data better.
+
+ - The GstVideoDecoder::handle_missing_data vfunc was added to
+ inform subclasses about packet loss or missing data and let them
+ handle it in their own way if they like.
+
+ - gst_video_decoder_set_needs_sync_point() lets subclasses signal
+ that they need the stream to start with a sync point. If
+ enabled, the base class will discard all non-sync point frames
+ in the beginning and after a flush and does not pass them to the
+ subclass. Furthermore, if the first frame is not a sync point,
+ the base class will try and request a sync frame from upstream
+ by sending a force-key-unit event (see next items).
+
+ - New "automatic-request-sync-points" and
+ "automatic-request-sync-point-flags" properties to automatically
+ request sync points when needed, e.g. on packet loss or if the
+ first frame is not a keyframe. Applications may want to enable
+ this on decoders operating in e.g. RTP/WebRTC/RTSP receiver
+ pipelines.
+
+ - The new "min-force-key-unit-interval" property can be used to
+ ensure there’s a minimal interval between keyframe requests to
+ upstream (and/or the sender) and we’re not flooding the sender
+ with key unit requests.
+
+ - gst_video_decoder_request_sync_point() allows subclasses to
+ request a new sync point (e.g. if they choose to do their own
+ missing data handling). This will still honour the
+ "min-force-key-unit-interval" property if set.
+
+Improved support for custom minimal GStreamer builds
+
+- Element registration and registration of other plugin features
+ inside plugin init functions has been improved in order to
+ facilitate minimal custom GStreamer builds.
+
+- A number of new macros have been added to declare and create
+ per-element and per-pluginfeature register functions in all plugins,
+ and then call those from the per-plugin plugin_init functions:
+
+ - GST_ELEMENT_REGISTER_DEFINE,
+ GST_DEVICE_PROVIDER_REGISTER_DEFINE,
+ GST_DYNAMIC_TYPE_REGISTER_DEFINE, GST_TYPE_FIND_REGISTER_DEFINE
+ for the actual registration call with GStreamer
+ - GST_ELEMENT_REGISTER, GST_DEVICE_PROVIDER_REGISTER,
+ GST_DYNAMIC_TYPE_REGISTER, GST_PLUGIN_STATIC_REGISTER,
+ GST_TYPE_FIND_REGISTER to call the registration function defined
+ by the REGISTER_DEFINE macro
+ - GST_ELEMENT_REGISTER_DECLARE,
+ GST_DEVICE_PROVIDER_REGISTER_DECLARE,
+ GST_DYNAMIC_TYPE_REGISTER_DECLARE,
+ GST_TYPE_FIND_REGISTER_DECLARE to declare the registration
+ function defined by the REGISTER_DEFINE macro
+ - and various variants for advanced use cases.
+
+- This means that applications can call the per-element and
+ per-pluginfeature registration functions for only the elements they
+ need instead of registering plugins as a whole with all kinds of
+ elements that may not be required (e.g. encoder and decoder instead
+ of just decoder). In case of static linking all unused functions and
+ their dependencies would be removed in this case by the linker,
+ which helps minimise binary size for custom builds.
+
+- gst_init() will automatically call a gst_init_static_plugins()
+ function if one exists.
+
+- See the GStreamer static build documentation and Stéphane’s blog
+ post Generate a minimal GStreamer build, tailored to your needs for
+ more details.
New elements
-- this section will be filled in in due course
+- New aesdec and aesenc elements for AES encryption and decryption in
+ a custom format.
+
+- New encodebin2 element with dynamic/sometimes source pads in order
+ to support the option of doing the muxing outside of encodebin,
+ e.g. in combination with a splitmuxsink.
+
+- New fakeaudiosink and videocodectestsink elements for testing and
+ debugging (see below for more details)
+
+- rtpisacpay, rtpisacdepay: new RTP payloader and depayloader for iSAC
+ audio codec
+
+- rtpst2022-1-fecdec, rtpst2022-1-fecenc: new elements providing SMPTE
+ 2022-1 2-D Forward Error Correction. More details in Mathieu’s blog
+ post.
+
+- isac: new plugin wrapping the Internet Speech Audio Codec reference
+ encoder and decoder from the WebRTC project.
+
+- asio: plugin for Steinberg ASIO (Audio Streaming Input/Output) API
+
+- gssrc, gssink: add source and sink for Google Cloud Storage
+
+- onnx: new plugin to apply ONNX neural network models to video
+
+- openaptx: aptX and aptX-HD codecs using libopenaptx (v0.2.0)
+
+- qroverlay, debugqroverlay: new elements that allows overlaying data
+ on top of video in form of a QR code
+
+- cvtracker: new OpenCV-based tracker element
+
+- av1parse, vp9parse: new parsers for AV1 and VP9 video
+
+- va: work on the new VA-API plugin implementation for
+ hardware-accelerated video decoding and encoding has continued at
+ pace, with various new decoders and filters having joined the
+ initial vah264dec:
+
+ - vah265dec: VA-API H.265 decoder
+ - vavp8dec: VA-API VP8 decoder
+ - vavp9dec: VA-API VP9 decoder
+ - vaav1dec: VA-API AV1 decoder
+ - vampeg2dec: VA-API MPEG-2 decoder
+ - vadeinterlace: : VA-API deinterlace filter
+ - vapostproc: : VA-API postproc filter (color conversion,
+ resizing, cropping, color balance, video rotation, skin tone
+ enhancement, denoise, sharpen)
+
+ See Víctor’s blog post “GstVA in GStreamer 1.20” for more details
+ and what’s coming up next.
+
+- vaapiav1dec: new AV1 decoder element (in gstreamer-vaapi)
+
+- msdkav1dec: hardware-accelerated AV1 decoder using the Intel Media
+ SDK / oneVPL
+
+- nvcodec plugin for NVIDIA NVCODEC API for hardware-accelerated video
+ encoding and decoding:
+
+ - cudaconvert, cudascale: new CUDA based video color space convert
+ and rescale elements
+ - cudaupload, cudadownload: new helper elements for memory
+ transfer between CUDA and system memory spaces
+ - nvvp8sldec, nvvp9sldec: new GstCodecs-based VP8/VP9 decoders
+
+- Various new hardware-accelerated elements for Windows:
+
+ - d3d11screencapturesrc: new desktop capture element, including a
+ GstDeviceProvider implementation to enumerate/select target
+ monitors for capture.
+ - d3d11av1dec and d3d11mpeg2dec: AV1 and MPEG-2 decoders
+ - d3d11deinterlace: deinterlacing filter
+ - d3d11compositor: video composing element
+ - see Windows section below for more details
+
+- new Rust plugins:
+
+ - audiornnoise: Removes noise from an audio stream
+ - awstranscribeparse: Parses AWS audio transcripts into timed text
+ buffers
+ - ccdetect: Detects if valid closed captions are present in a
+ closed captions stream
+ - cea608tojson: Converts CEA-608 Closed Captions to a JSON
+ representation
+ - cmafmux: CMAF fragmented MP4 muxer
+ - dashmp4mux: DASH fragmented MP4 muxer
+ - isofmp4mux: ISO fragmented MP4 muxer
+ - ebur128level: EBU R128 Loudness Level Measurement
+ - ffv1dec: FFV1 video decoder
+ - gtk4paintablesink: GTK4 video sink, which provides a
+ GdkPaintable that can be rendered in various widgets
+ - hlssink3: HTTP Live Streaming sink
+ - hrtfrender: Head-Related Transfer Function (HRTF) renderer
+ - hsvdetector: HSV colorspace detector
+ - hsvfilter: HSV colorspace filter
+ - jsongstenc: Wraps buffers containing any valid top-level JSON
+ structures into higher level JSON objects, and outputs those as
+ ndjson
+ - jsongstparse: Parses ndjson as output by jsongstenc
+ - jsontovtt: converts JSON to WebVTT subtitles
+ - regex: Applies regular expression operations on text
+ - roundedcorners: Adds rounded corners to video
+ - spotifyaudiosrc: Spotify source
+ - textahead: Display upcoming text buffers ahead (e.g. for
+ Karaoke)
+ - transcriberbin: passthrough bin that transcribes raw audio to
+ closed captions using awstranscriber and puts the captions as
+ metas onto the video
+ - tttojson: Converts timed text to a JSON representation
+ - uriplaylistbin: Playlist source bin
+ - webpdec-rs: WebP image decoder with animation support
+
+- New plugin codecalpha with elements to assist with WebM Alpha
+ decoding
+
+ - codecalphademux: Split stream with GstVideoCodecAlphaMeta into
+ two streams
+ - alphacombine: Combine two raw video stream (I420 or NV12) as one
+ stream with alpha channel (A420 or AV12)
+ - vp8alphadecodebin: A bin to handle software decoding of VP8 with
+ alpha
+ - vp9alphadecodebin: A bin to handle software decoding of VP9 with
+ alpha
+
+- New hardware accelerated elements for Linux:
+
+ - v4l2slmpeg2dec: Support for Linux Stateless MPEG2 decoders
+ - v4l2slvp9dec: Support for Linux Stateless VP9 decoders
+ - v4l2slvp8alphadecodebin: Support HW accelerated VP8 with alpha
+ layer decoding
+ - v4l2slvp9alphadecodebin: Support HW accelerated VP9 with alpha
+ layer decoding
New element features and additions
-- this section will be filled in in due course
+- assrender: handle more font mime types; better interaction with
+ matroskademux for embedded fonts
+
+- audiobuffersplit: Add support for specifying output buffer size in
+ bytes (not just duration)
+
+- audiolatency: new "samplesperbuffer" property so users can configure
+ the number of samples per buffer. The default value is 240 samples
+ which is equivalent to 5ms latency with a sample rate of 48000,
+ which might be larger than actual buffer size of audio capture
+ device.
+
+- audiomixer, audiointerleave, GstAudioAggregator: now keep a count of
+ samples that are dropped or processed as statistic and can be made
+ to post QoS messages on the bus whenever samples are dropped by
+ setting the "qos-messages" property on input pads.
+
+- audiomixer, compositor: improved handling of new inputs added at
+ runtime. New API was added to the GstAggregator base class to allow
+ subclasses to opt into an aggregation mode where inactive pads are
+ ignored when processing input buffers
+ (gst_aggregator_set_ignore_inactive_pads(),
+ gst_aggregator_pad_is_inactive()). An “inactive pad” in this context
+ is a pad which, in live mode, hasn’t yet received a first buffer,
+ but has been waited on at least once. What would happen usually in
+ this case is that the aggregator would wait for data on this pad
+ every time, up to the maximum configured latency. This would
+ inadvertently push mixer elements in live mode to the configured
+ latency envelope and delay processing when new inputs are added at
+ runtime until these inputs have actually produced data. This is
+ usually undesirable. With this new API, new inputs can be added
+ (requested) and configured and they won’t delay the data processing.
+ Applications can opt into this new behaviour by setting the
+ "ignore-inactive-pads" property on compositor, audiomixer or other
+ GstAudioAggregator-based elements.
+
+- cccombiner: implement “scheduling” of captions. So far cccombiner’s
+ behaviour was essentially that of a funnel: it strictly looked at
+ input timestamps to associate together video and caption buffers.
+ Now it will try to smoothly schedule caption buffers in order to
+ have exactly one per output video buffer. This might involve
+ rewriting input captions, for example when the input is CDP then
+ sequence counters are rewritten, time codes are dropped and
+ potentially re-injected if the input video frame had a time code
+ meta. This can also lead to the input drifting from synchronization,
+ when there isn’t enough padding in the input stream to catch up. In
+ that case the element will start dropping old caption buffers once
+ the number of buffers in its internal queue reaches a certain limit
+ (configurable via the "max-scheduled" property). The new original
+ funnel-like behaviour can be restored by setting the "scheduling"
+ property to FALSE.
+
+- ccconverter: new "cdp-mode" property to specify which sections to
+ include in CDP packets (timecode, CC data, service info). Various
+ software, including ffmpeg’s Decklink support, fails parsing CDP
+ packets that contain anything but CC data in the CDP packets.
+
+- clocksync: new "sync-to-first" property for automatic timestamp
+ offset setup: if set clocksync will set up the "ts-offset" value
+ based on the first buffer and the pipeline’s running time when the
+ first buffer arrived. The newly configured "ts-offset" in this case
+ would be the value that allows outputting the first buffer without
+ waiting on the clock. This is useful for example to feed a non-live
+ input into an already-running pipeline.
+
+- compositor:
+
+ - multi-threaded input conversion and compositing. Set the
+ "max-threads" property to activate this.
+ - new "sizing-policy" property to support display aspect ratio
+ (DAR)-aware scaling. By default the image is scaled to fill the
+ configured destination rectangle without padding and without
+ keeping the aspect ratio. With sizing-policy=keep-aspect-ratio
+ the input image is scaled to fit the destination rectangle
+ specified by GstCompositorPad:{xpos, ypos, width, height}
+ properties preserving the aspect ratio. As a result, the image
+ will be centered in the destination rectangle with padding if
+ necessary.
+ - new "zero-size-is-unscaled" property on input pads. By default
+ pad width=0 or pad height=0 mean that the stream should not be
+ scaled in that dimension. But if the "zero-size-is-unscaled"
+ property is set to FALSE a width or height of 0 is instead
+ interpreted to mean that the input image on that pad should not
+ be composited, which is useful when creating animations where an
+ input image is made smaller and smaller until it disappears.
+ - improved handling of new inputs at runtime via
+ "ignore-inactive-pads"property (see above for details)
+ - allow output format with alpha even if none of the inputs have
+ alpha (also glvideomixer and other GstVideoAggregator
+ subclasses)
+
+- dashsink: add h265 codec support and signals for allowing custom
+ playlist/fragment output
+
+- decodebin3:
+
+ - improved decoder selection, especially for hardware decoders
+ - make input activation “atomic” when adding inputs dynamically
+ - better interleave handling: take into account decoder latency
+ for interleave size
+
+- decklink:
+
+ - Updated DeckLink SDK to 11.2 to support DeckLink 8K Pro
+ - decklinkvideosrc:
+ - More accurate and stable capture timestamps: use the
+ hardware reference clock time when the frame was finished
+ being captured instead of a clock time much further down the
+ road.
+ - Automatically detect widescreen vs. normal NTSC/PAL
+
+- encodebin:
+
+ - add “smart encoding” support for H.265, VP8 and VP9 (i.e. only
+ re-encode where needed and otherwise pass through encoded video
+ as-is).
+ - H264/H265 smart encoding improvements: respect user-specified
+ stream-format, but if not specified default to avc3/hvc1 with
+ in-band SPS/PPS/VPS signalling for more flexibility.
+ - new encodebin2 element with dynamic/sometimes source pads in
+ order to support the option of doing the muxing outside of
+ encodebin, e.g. in combination with splitmuxsink.
+ - add APIs to set element properties on encoding profiles (see
+ below)
+
+- errorignore: new "ignore-eos" property to also ignore FLOW_EOS from
+ downstream elements
+
+- giosrc: add support for growing source files: applications can
+ specify that the underlying file being read is growing by setting
+ the "is-growing" property. If set, the source won’t EOS when it
+ reaches the end of the file, but will instead start monitoring it
+ and will start reading data again whenever a change is detected. The
+ new "waiting-data" and "done-waiting-data" signals keep the
+ application informed about the current state.
+
+- gtksink, gtkglsink:
+
+ - scroll event support: forwarded as navigation events into the
+ pipeline
+ - "video-aspect-ratio-override" property to force a specific
+ aspect ratio
+ - "rotate-method" property and support automatic rotation based on
+ image tags
+
+- identity: new "stats" property allows applications to retrieve the
+ number of bytes and buffers that have passed through so far.
+
+- interlace: add support for more formats, esp 10-bit, 12-bit and
+ 16-bit ones
+
+- jack: new "low-latency" property for automatic latency-optimized
+ setting and "port-names" property to select ports explicitly
+
+- jpegdec: support output conversion to RGB using libjpeg-turbo (for
+ certain input files)
+
+- line21dec:
+
+ - "mode" property to control whether and how detected closed
+ captions should be inserted in the list of existing close
+ caption metas on the input frame (if any): add, drop, or
+ replace.
+ - "ntsc-only" property to only look for captions if video has NTSC
+ resolution
+
+- line21enc: new "remove-caption-meta" to remove metas from output
+ buffers after encoding the captions into the video data; support for
+ CDP closed captions
+
+- matroskademux, matroskamux: Add support for ffv1, a lossless
+ intra-frame video coding format.
+
+- matroskamux: accept in-band SPS/PPS/VPS for H264 and H265
+ (i.e. stream-format avc3 and hev1) which allows on-the-fly
+ profile/level/resolution changes.
+
+- matroskamux: new "cluster-timestamp-offset" property, useful for use
+ cases where the container timestamps should map to some absolute
+ wall clock time, for example.
+
+- rtpsrc: add "caps" property to allow explicit setting of the caps
+ where needed
+
+- mpegts: support SCTE-35 passthrough via new "send-scte35-events"
+ property on MPEG-TS demuxer tsdemux. When enabled, SCTE 35 sections
+ (eg ad placement opportunities) are forwarded as events donwstream
+ where they can be picked up again by mpegtsmux. This required a
+ semantic change in the SCTE-35 section API: timestamps are now in
+ running time instead of muxer pts.
+
+- tsdemux: Handle PCR-less MPEG-TS streams; more robust timestamp
+ handling in certain corner cases and for poorly muxed streams.
+
+- mpegtsmux:
+
+ - More conformance improvements to make MPEG-TS analyzers happy:
+ - PCR timing accuracy: Improvements to the way mpegtsmux
+ outputs PCR observations in CBR mode, so that a PCR
+ observation is always inserted when needed, so that we never
+ miss the configured pcr-interval, as that triggers various
+ MPEG-TS analyser errors.
+ - Improved PCR/SI scheduling
+ - Don’t write PCR until PAT/PMT are output to make sure streams
+ start cleanly with a PAT/PMT.
+ - Allow overriding the automatic PMT PID selection via
+ application-supplied PMT_%d fields in the prog-map
+ structure/property.
+
+- mp4mux:
+
+ - new "first-moov-then-finalise" mode for fragmented output where
+ the output will start with a self-contained moov atom for the
+ first fragment, and then produce regular fragments. Then at the
+ end when the file is finalised, the initial moov is invalidated
+ and a new moov is written covering the entire file. This way the
+ file is a “fragmented mp4” file while it is still being written
+ out, and remains playable at all times, but at the end it is
+ turned into a regular mp4 file (with former fragment headers
+ remaining as unused junk data in the file).
+ - support H.264 avc3 and H.265 hvc1 stream formats as input where
+ the codec data is signalled in-band inside the bitstream instead
+ of caps/file headers.
+ - support profile/level/resolution changes for H264/H265 input
+ streams (i.e. codec data changing on the fly). Each codec_data
+ is put into its own SampleTableEntry inside the stsd, unless the
+ input is in avc3 stream format in which case it’s written
+ in-band an not in the headers.
+
+- multifilesink: new ""min-keyframe-distance"" property to make
+ minimum distance between keyframes in next-file=key-frame mode
+ configurable instead of hard-coding it to 10 seconds.
+
+- mxfdemux has seen a big refactoring to support non-frame wrappings
+ and more accurate timestamp/seek handling for some formats
+
+- msdk plugin for hardware-accelerated video encoding and decoding
+ using the Intel Media SDK:
+
+ - oneVPL support (Intel oneAPI Video Processing Library)
+ - AV1 decoding support
+ - H.264 decoder now supports constrained-high and progressive-high
+ profiles
+ - H.264 encoder:
+ - more configuration options (properties):
+ "intra-refresh-type", "min-qp" , "max-qp", "p-pyramid",
+ "dblk-idc"
+ - H.265 encoder:
+ - can output main-still-picture profile
+ - now inserts HDR SEIs (mastering display colour volume and
+ content light level)
+ - more configuration options (properties):
+ "intra-refresh-type", "min-qp" , "max-qp", "p-pyramid",
+ "b-pyramid", "dblk-idc", "transform-skip"
+ - support for RGB 10bit format
+ - External bitrate control in encoders
+ - Video post proc element msdkvpp gained support for 12-bit pixel
+ formats P012_LE, Y212_LE and Y412_LE
+
+- nvh264sldec: interlaced stream support
+
+- openh264enc: support main, high, constrained-high and
+ progressive-high profiles
+
+- openjpeg: support for multithreaded decoding and encoding
+
+- rtspsrc: now supports IPv6 also for tunneled mode (RTSP-over-HTTP);
+ new "ignore-x-server-reply" property to ignore the
+ x-server-ip-address server header reply in case of HTTP tunneling,
+ as it is often broken.
+
+- souphttpsrc: Runtime compatibility support for libsoup2 and
+ libsoup3. libsoup3 is the latest major version of libsoup, but
+ libsoup2 and libsoup3 can’t co-exist in the same process because
+ there is no namespacing or versioning for GObject types. As a
+ result, it would be awkward if the GStreamer souphttpsrc plugin
+ linked to a specific version of libsoup, because it would only work
+ with applications that use the same version of libsoup. To make this
+ work, the soup plugin now tries to determine the libsoup version
+ used by the application (and its other dependencies) at runtime on
+ systems where GStreamer is linked dynamically. libsoup3 support is
+ still considered somewhat experimental at this point.
+
+- srtsrc, srtsink: add signals for the application to accept/reject
+ incoming connections
+
+- timeoverlay: new elapsed-running-time time mode which shows the
+ running time since the first running time (and each flush-stop).
+
+- udpsrc: new timestamping mode to retrieve packet receive timestamps
+ from the kernel via socket control messages (SO_TIMESTAMPNS) on
+ supported platforms
+
+- uritranscodebin: new setup-source and element-setup signals for
+ applications to configure elements used
+
+- v4l2codecs plugin gained support for 4x4 and 32x32 tile formats
+ enabling some platforms or direct renders. Important memory usage
+ improvement.
+
+- v4l2slh264dec now implements the final Linux uAPI as shipped on
+ Linux 5.11 and later.
+
+- valve: add "drop-mode" property and provide two new modes of
+ operation: in drop-mode=forward-sticky-events sticky events
+ (stream-start, segment, tags, caps, etc.) are forwarded downstream
+ even when dropping is enabled; drop-mode=transform-to-gap will in
+ addition also convert buffers into gap events when dropping is
+ enabled, which lets downstream elements know that time is advancing
+ and might allow for preroll in many scenarios. By default all events
+ and all buffers are dropped when dropping is enabled, which can
+ cause problems with caps negotiation not progressing or branches not
+ prerolling when dropping is enabled.
+
+- videocrop: support for many more pixel formats, e.g. planar YUV
+ formats with > 8bits and GBR* video formats; can now also accept
+ video not backed by system memory as long as downstream supports the
+ GstCropMeta
+
+- videotestsrc: new smpte-rp-219 pattern for SMPTE75 RP-219 conformant
+ color bars
+
+- vp8enc: finish support for temporal scalability: two new properties
+ ("temporal-scalability-layer-flags",
+ "temporal-scalability-layer-sync-flags") and a unit change on the
+ "temporal-scalability-target-bitrate" property (now expects bps);
+ also make temporal scalability details available to RTP payloaders
+ as buffer metadata.
+
+- vp9enc: new properties to tweak encoder performance:
+
+ - "aq-mode" to configure adaptive quantization modes
+ - "frame-parallel-decoding" to configure whether to create a
+ bitstream that reduces decoding dependencies between frames
+ which allows staged parallel processing of more than one video
+ frames in the decoder. (Defaults to TRUE)
+ - "row-mt", "tile-columns" and "tile-rows" so multithreading can
+ be enabled on a per-tile basis, instead of on a per tile-column
+ basis. In combination with the new "tile-rows" property, this
+ allows the encoder to make much better use of the available CPU
+ power.
+
+- vp9dec, vp9enc: add support for 10-bit 4:2:0 and 4:2:2 YUV, as well
+ as 8-bit 4:4:4
+
+- vp8enc, vp9enc now default to “good quality” for the deadline
+ property rather then “best quality”. Having the deadline set to best
+ quality causes the encoder to be absurdly slow, most real-life users
+ will prefer good-enough quality with better performance instead.
+
+- wpesrc:
+
+ - implement audio support: a new sometimes source pad will be
+ created for each audio stream created by the web engine.
+ - move wpesrc to wpevideosrc and add a wrapper bin wpesrc to also
+ support audio
+ - also handles web:// URIs now (same as cefsrc)
+ - post messages with the estimated load progress on the bus
-Plugin and library moves
+- x265enc: add negative DTS support, which means timestamps are now
+ offset by 1h same as with x264enc
+
+RTP Payloaders and Depayloaders
+
+- rtpisacpay, rtpisacdepay: new RTP payloader and depayloader for iSAC
+ audio codec
+
+- rtph264depay:
+
+ - new "request-keyframe" property to make the depayloader
+ automatically request a new keyframe from the sender on packet
+ loss, consistent with the new property on rtpvp8depay.
+ - new "wait-for-keyframe" property to make depayloader wait for a
+ new keyframe at the beginning and after packet loss (only
+ effective if the depayloader outputs AUs), consistent with the
+ existing property on rtpvp8depay.
+
+- rtpopuspay, rtpopusdepay: support libwebrtc-compatible multichannel
+ audio in addition to the previously supported multichannel audio
+ modes
+
+- rtpopuspay: add DTX (Discontinuous Transmission) support
+
+- rtpvp8depay: new "request-keyframe" property to make the depayloader
+ automatically request a new keyframe from the sender on packet loss.
+
+- rtpvp8pay: temporal scaling support
-- this section will be filled in in due course
+- rtpvp9depay: Improved SVC handling (aggregate all layers)
+
+RTP Infrastructure
+
+- rtpst2022-1-fecdec, rtpst2022-1-fecenc: new elements providing SMPTE
+ 2022-1 2-D Forward Error Correction. More details in Mathieu’s blog
+ post.
+
+- rtpreddec: BUNDLE support
+
+- rtpredenc, rtpulpfecenc: add support for Transport-wide Congestion
+ Control (TWCC)
+
+- rtpsession: new "twcc-feedback-interval" property to allow RTCP TWCC
+ reports to be scheduled on a timer instead of per marker-bit.
+
+Plugin and library moves
- There were no plugin moves or library moves in this cycle.
The following elements or plugins have been removed:
-- this section will be filled in in due course
-
- The ofa audio fingerprinting plugin has been removed. The MusicIP
database has been defunct for years so this plugin is likely neither
useful nor used by anyone.
Miscellaneous API additions
-- this section will be filled in in due course
+Core
+
+- gst_buffer_new_memdup() is a convenience function for the
+ widely-used gst_buffer_new_wrapped(g_memdup(data,size),size)
+ pattern.
+
+- gst_caps_features_new_single() creates a new single GstCapsFeatures,
+ avoiding the need to use the vararg function with NULL terminator
+ for simple cases.
+
+- gst_element_type_set_skip_documentation() can be used by plugins to
+ signal that certain elements should not be included in the GStreamer
+ plugin documentation. This is useful for plugins where elements are
+ registered dynamically based on hardware capabilities and/or where
+ the available plugins and properties vary from system to system.
+ This is used in the d3d11 plugin for example to ensure that only the
+ list of default elements is advertised in the documentation.
+
+- gst_type_find_suggest_empty_simple() is a new convenience function
+ for typefinders for cases where there’s only a media type and no
+ other fields.
+
+- New API to create elements and set properties at construction time,
+ which is not only convenient, but also allows GStreamer elements to
+ have construct-only properties: gst_element_factory_make_full(),
+ gst_element_factory_make_valist(),
+ gst_element_factory_make_with_properties(),
+ gst_element_factory_create_full(),
+ gst_element_factory_create_valist(),
+ gst_element_factory_create_with_properties().
+
+- GstSharedTaskPool: new “shared” task pool subclass with slightly
+ different default behaviour than the existing GstTaskPool which
+ would create unlimited number of threads for new tasks. The shared
+ taskpool creates up to N threads (default: 1) and then distributes
+ pending tasks to those threads round-robin style, and blocks if no
+ thread is available. It is possible to join tasks. This can be used
+ by plugins to implement simple multi-threaded processing and is used
+ for the new multi-threaded video conversion and compositing done in
+ GstVideoAggregator, videoconverter and compositor.
+
+Plugins Base Utils library
+
+- GstDiscoverer:
+
+ - gst_discoverer_container_info_get_tags() was added to retrieve
+ global/container tags (vs. per-stream tags). Per-Stream tags can
+ be retrieved via the existing
+ gst_discoverer_stream_info_get_tags().
+ gst_discoverer_info_get_tags(), which for many files returns a
+ confusing mix of stream and container tags, has been deprecated
+ in favour of the container/stream-specific functions.
+ - gst_discoverer_stream_info_get_stream_number() returns a unique
+ integer identifier for a given stream within the given
+ GstDiscoverer context. (If this matches the stream number inside
+ the container bitstream that’s by coincidence and not by
+ design.)
+
+- gst_pb_utils_get_caps_description_flags() can be used to query
+ whether certain caps represent a container, audio, video, image,
+ subtitles, tags, or something else. This only works for formats
+ known to GStreamer.
+
+- gst_pb_utils_get_file_extension_from_caps() returns a possible file
+ extension for given caps.
+
+- gst_codec_utils_h264_get_profile_flags_level(): Parses profile,
+ flags, and level from H264 AvcC codec_data. The format of H264 AVCC
+ extradata/sequence_header is documented in the ITU-T H.264
+ specification section 7.3.2.1.1 as well as in ISO/IEC 14496-15
+ section 5.3.3.1.2.
+
+- gst_codec_utils_caps_get_mime_codec() to convert caps to a RFC 6381
+ compatible MIME codec string codec. Useful for providing the codecs
+ field inside the Content-Type HTTP header for containerized formats,
+ such as mp4 or matroska.
+
+GStreamer OpenGL integration library and plugins
+
+- glcolorconvert: added suppport for converting the video formats
+ A420, AV12, BGR, BGRA, RGBP and BGRP.
+
+- Added support to GstGLBuffer for persistent buffer mappings where a
+ Pixel Buffer Object (PBO) can be mapped by both the CPU and the GPU.
+ This removes a memcpy() when uploading textures or vertices
+ particularly when software decoders (e.g. libav) are direct
+ rendering into our memory. Improves transfer performance
+ significantly. Requires OpenGL 4.4, GL_ARB_buffer_storage or
+ GL_EXT_buffer_storage
+
+- Added various helper functions for handling 4x4 matrices of affine
+ transformations as used by GstVideoAffineTransformationMeta.
+
+- Add support to GstGLContext for allowing the application to control
+ the config (EGLConfig, GLXConfig, etc) used when creating the OpenGL
+ context. This allows the ability to choose between RGB16 or RGB10A2
+ or RGBA8 back/front buffer configurations that were previously
+ hardcoded. GstGLContext also supports retrieving the configuration
+ it was created with or from an externally provide OpenGL context
+ handle. This infrastructure is also used to create a compatible
+ config from an application/externally provided OpenGL context in
+ order to improve compatibility with other OpenGL frameworks and GUI
+ toolkits. A new environment variable GST_GL_CONFIG was also added to
+ be able to request a specific configuration from the command line.
+ Note: different platforms will have different functionality
+ available.
+
+- Add support for choosing between EGL and WGL at runtime when running
+ on Windows. Previously this was a build-time switch. Allows use in
+ e.g. Gtk applications on Windows that target EGL/ANGLE without
+ recompiling GStreamer. gst_gl_display_new_with_type() can be used by
+ applications to choose a specific display type to use.
+
+- Build fixes to explicitly check for Broadcom-specific libraries on
+ older versions of the Raspberry Pi platform. The Broadcom OpenGL ES
+ and EGL libraries have different filenames. Using the vc4 Mesa
+ driver on the Raspberry Pi is not affected.
+
+- Added support to glupload and gldownload for transferring RGBA
+ buffers using the memory:NVMM available on the Nvidia Tegra family
+ of embedded devices.
+
+- Added support for choosing libOpenGL and libGLX as used in a GLVND
+ environment on unix-based platforms. This allows using desktop
+ OpenGL and EGL without pulling in any GLX symbols as would be
+ required with libGL.
+
+Video library
+
+- New raw video formats:
+
+ - AV12 (NV12 with alpha plane)
+ - RGBP and BGRP (planar RGB formats)
+ - ARGB64 variants with specified endianness instead of host
+ endianness:
+ - ARGB64_LE, ARGB64_BE
+ - RGBA64_BE, RGBA64_LE
+ - BGRA64_BE, BGRA64_LE
+ - ABGR64_BE, ABGR64_LE
+
+- gst_video_orientation_from_tag() is new convenience API to parse the
+ image orientation from a GstTagList.
+
+- GstVideoDecoder subframe support (see below)
+
+- GstVideoCodecState now also carries some HDR metadata
+
+- Ancillary video data: implement transform functions for AFD/Bar
+ metas, so they will be forwarded in more cases
+
+MPEG-TS library
+
+This library only handles section parsing and such, see above for
+changes to the actual mpegtsmux and mpegtsdemux elements.
+
+- many additions and improvements to SCTE-35 section parsing
+- new API for fetching extended descriptors:
+ gst_mpegts_find_descriptor_with_extension()
+- add support for SIT sections (Selection Information Tables)
+- expose event-from-section constructor gst_event_new_mpegts_section()
+- parse Audio Preselection Descriptor needed for Dolby AC-4
+
+GstWebRTC library + webrtcbin
+
+- Change the way in which sink pads and transceivers are matched
+ together to support easier usage. If a pad is created without a
+ specific index (i.e. using sink_%u as the pad template), then an
+ available compatible transceiver will be searched for. If a specific
+ index is requested (i.e. sink_1) then if a transceiver for that
+ m-line already exists, that transceiver must match the new sink pad
+ request. If there is no transceiver available in either scenario, a
+ new transceiver is created. If a mixture of both sink_1 and sink_%u
+ requests result in an impossible situation, an error will be
+ produced at pad request time or from create offer/answer.
+
+- webrtcbin now uses regular ICE nomination instead of libnice’s
+ default of aggressive ICE nomination. Regular ICE nomination is the
+ default recommended by various relevant standards and improves
+ connectivity in specific network scenarios.
+
+- Add support for limiting the port range used for RTP with the
+ addition of the min-rtp-port and max-rtp-port properties on the ICE
+ object.
+
+- Expose the SCTP transport as a property on webrtcbin to more closely
+ match the WebRTC specification.
+
+- Added support for taking into account the data channel transport
+ state when determining the value of the "connection-state" property.
+ Previous versions of the WebRTC spec did not include the data
+ channel state when computing this value.
+
+- Add configuration for choosing the size of the underlying sockets
+ used for transporting media data
+
+- Always advertise support for the transport-cc RTCP feedback protocol
+ as rtpbin supports it. For full support, the configured caps (input
+ or through codec-preferences) need to include the relevant RTP
+ header extension.
+
+- Numerous fixes to caps and media handling to fail-fast when an
+ incompatible situation is detected.
+
+- Improved support for attaching the required media after a remote
+ offer has been set.
+
+- Add support for dynamically changing the amount of FEC used for a
+ particular stream.
+
+- webrtcbin now stops further SDP processing at the first error it
+ encounters.
+
+- Completed support for either local or the remote closing a data
+ channel.
+
+- Various fixes when performing BUNDLEing of the media streams in
+ relation to RTX and FEC usage.
+
+- Add support for writing out QoS DSCP marking on outgoing packets to
+ improve reliability in some network scenarios.
+
+- Improvements to the statistics returned by the get-stats signal
+ including the addition of the raw statistics from the internal
+ RTPSource, the TWCC stats when available.
+
+- The webrtc library does not expose any objects anymore with public
+ fields. Instead properties have been added to replace that
+ functionality. If you are accessing such fields in your application,
+ switch to the corresponding properties.
+
+GstCodecs and Video Parsers
+
+- Support for render delays to improve throughput across all CODECs
+ (used with NVDEC and V4L2).
+- lots of improvements to parsers and the codec parsing decoder base
+ classes (H264, H265, VP8, VP9, AV1, MPEG-2) used for various
+ hardware-accelerate decoder APIs.
+
+Bindings support
+
+- gst_allocation_params_new() allocates a GstAllocationParams struct
+ on the heap. This should only be used by bindings (and freed via
+ gst_allocation_params_free() then). In C code you would allocate
+ this on the stack and only init it in place.
+
+- gst_debug_log_literal() can be used to log a string to the debug log
+ without going through any printf format expansion and associated
+ overhead. This is mostly useful for bindings such as the Rust
+ bindings which may have done their own formatting already .
+
+- Provide non-inlined versions of refcounting APIs for various
+ GStreamer mini objects, so that they can be consumed by bindings
+ (e.g. gstreamer-sharp): gst_buffer_ref, gst_buffer_unref,
+ gst_clear_buffer, gst_buffer_copy, gst_buffer_replace,
+ gst_buffer_list_ref, gst_buffer_list_unref, gst_clear_buffer_list,
+ gst_buffer_list_copy, gst_buffer_list_replace, gst_buffer_list_take,
+ gst_caps_ref, gst_caps_unref, gst_clear_caps, gst_caps_replace,
+ gst_caps_take, gst_context_ref, gst_context_unref, gst_context_copy,
+ gst_context_replace, gst_event_replace, gst_event_steal,
+ gst_event_take, gst_event_ref, gst_event_unref, gst_clear_event,
+ gst_event_copy, gst_memory_ref, gst_memory_unref, gst_message_ref,
+ gst_message_unref, gst_clear_message, gst_message_copy,
+ gst_message_replace, gst_message_take, gst_promise_ref,
+ gst_promise_unref, gst_query_ref, gst_query_unref, gst_clear_query,
+ gst_query_copy, gst_query_replace, gst_query_take, gst_sample_ref,
+ gst_sample_unref, gst_sample_copy, gst_tag_list_ref,
+ gst_tag_list_unref, gst_clear_tag_list, gst_tag_list_replace,
+ gst_tag_list_take, gst_uri_copy, gst_uri_ref, gst_uri_unref,
+ gst_clear_uri.
+
+- expose a GType for GstMiniObject
+
+- gst_device_provider_probe() now returns non-floating device object
+
+API Deprecations
+
+- gst_element_get_request_pad() has been deprecated in favour of the
+ newly-added gst_element_request_pad_simple() which does the exact
+ same thing but has a less confusing name that hopefully makes clear
+ that the function request a new pad rather than just retrieves an
+ already-existing request pad.
+
+- gst_discoverer_info_get_tags(), which for many files returns a
+ confusing mix of stream and container tags, has been deprecated in
+ favour of the container-specific and stream-specific functions,
+ gst_discoverer_container_info_get_tags() and
+ gst_discoverer_stream_info_get_tags().
+
+- gst_video_sink_center_rect() was deprecated in favour of the more
+ generic newly-added gst_video_center_rect().
+
+- The GST_MEMORY_FLAG_NO_SHARE flag has been deprecated, as it tends
+ to cause problems and prevents sub-buffering. If pooling or lifetime
+ tracking is required, memories should be allocated through a custom
+ GstAllocator instead of relying on the lifetime of the buffers the
+ memories were originally attached to, which is fragile anyway.
+
+- The GstPlayer high-level playback library is being replaced with the
+ new GstPlay library (see above). GstPlayer should be considered
+ deprecated at this point and will be marked as such in the next
+ development cycle. Applications should be ported to GstPlay.
+
+- Gstreamer Editing Services: ges_video_transition_set_border(),
+ ges_video_transition_get_border()
+ ges_video_transition_set_inverted()
+ ges_video_transition_is_inverted() have been deprecated, use
+ ges_timeline_element_set_children_properties() instead.
Miscellaneous performance, latency and memory optimisations
-- this section will be filled in in due course
+More video conversion fast paths
-Miscellaneous other changes and enhancements
+- v210 ↔ I420, YV12, Y42B, UYVY and YUY2
+- A420 → RGB
+
+Less jitter when waiting on the system clock
+
+- Better system clock wait accuracy, less jitter: where available,
+ clock_nanosleep is used for higher accuracy for waits below 500
+ usecs, and waits below 2ms will first use the regular waiting system
+ and then clock_nanosleep for the remainder. The various wait
+ implementation have a latency ranging from 50 to 500+ microseconds.
+ While this is not a major issue when dealing with a low number of
+ waits per second (for ex: video), it does introduce a non-negligible
+ jitter for synchronisation of higher packet rate systems.
+
+Video decoder subframe support
-- this section will be filled in in due course
+- The GstVideoDecoder base class gained API to process input at the
+ sub-frame level. That way video decoders can start decoding slices
+ before they have received the full input frame in its entirety (to
+ the extent this is supported by the codec, of course). This helps
+ with CPU utilisation and reduces latency.
-Tracing framework and debugging improvements
+- This functionality is now being used in the OpenJPEG JPEG 2000
+ decoder, the FFmpeg H.264 decoder (in case of NAL-aligned input) and
+ the OpenMAX H.264/H.265 decoders (in case of NAL-aligned input).
-- this section will be filled in in due course
+Miscellaneous other changes and enhancements
+
+- GstDeviceMonitor no longer fails to start just because one of the
+ device providers failed to start. That could happen for example on
+ systems where the pulseaudio device provider is installed, but
+ pulseaudio isn’t actually running but ALSA is used for audio
+ instead. In the same vein the device monitor now keeps track of
+ which providers have been started (via the new
+ gst_device_provider_is_started()) and only stops actually running
+ device providers when stopping the device monitor.
+
+- On embedded systems it can be useful to create a registry that can
+ be shared and read by multiple processes running as different users.
+ It is now possible to set the new GST_REGISTRY_MODE environment
+ variable to specify the file mode for the registry file, which by
+ default is set to be only user readable/writable.
+
+- GstNetClientClock will signal lost sync in case the remote time
+ resets (e.g. because device power cycles), by emitting the “synced”
+ signal with synced=FALSE parameter, so applications can take action.
+
+- gst_value_deserialize_with_pspec() allows deserialization with a
+ hint for what the target GType should be. This allows for example
+ passing arrays of flags through the command line or
+ gst_util_set_object_arg(), eg: foo="<bar,bar+baz>".
+
+- It’s now allowed to create an empty GstVideoOverlayComposition
+ without any rectangles by passing a NULL rectangle to
+ gst_video_overlay_composition_new(). This is useful for bindings and
+ simplifies application code in some places.
+
+Tracing framework, debugging and testing improvements
+
+- New factories tracer to list loaded elements (and other plugin
+ features). This can be useful to collect a list of elements needed
+ for an application, which then in turn can be used to create a
+ tailored minimal GStreamer build that contains just the elements
+ needed and nothing else.
+- New plugin-feature-loaded tracing hook for use by tracers like the
+ new factories tracer
+
+- GstHarness: Add gst_harness_set_live() so that harnesses can be set
+ to non-live and return is-live=false in latency queries if needed.
+ Default behaviour is to always return is-live=true in latency
+ queries.
+
+- navseek: new "hold-eos" property. When enabled, the element will
+ hold back an EOS event until the next keystroke (via navigation
+ events). This can be used to keep a video sink showing the last
+ frame of a video pipeline until a key is pressed instead of tearing
+ it down immediately on EOS.
+
+- New fakeaudiosink element: mimics an audio sink and can be used for
+ testing and CI pipelines on systems where no audio system is
+ installed or running. It differs from fakesink in that it only
+ support audio caps and syncs to the clock by default like a normal
+ audio sink. It also implements the GstStreamVolume interface like
+ most audio sinks do.
+
+- New videocodectestsink element for video codec conformance testing:
+ Calculates MD5 checksums for video frames and skips any padding
+ whilst doing so. Can optionally also write back the video data with
+ padding removed into a file for easy byte-by-byte comparison with
+ reference data.
Tools
-- this section will be filled in in due course
+gst-inspect-1.0
+
+- Can sort the list of plugins by passing --sort=name as command line
+ option
+
+gst-launch-1.0
+
+- will now error out on top-level properties that don’t exist and
+ which were silently ignored before
+- On Windows the high-resolution clock is enabled now, which provides
+ better clock and timer performance on Windows (see Windows section
+ below for more details).
+
+gst-play-1.0
+
+- New --start-position command line argument to start playback from
+ the specified position
+- Audio can be muted/unmuted in interactive mode by pressing the m
+ key.
+- On Windows the high-resolution clock is enabled now (see Windows
+ section below for more details)
+
+gst-device-monitor-1.0
+
+- New --include-hidden command line argument to also show “hidden”
+ device providers
+
+ges-launch-1.0
+
+- New interactive mode that allows seeking and such. Can be disabled
+ by passing the --no-interactive argument on the command line.
+- Option to forward tags
+- Allow using an existing clip to determine the rendering format (both
+ topology and profile) via new --profile-from command line argument.
GStreamer RTSP server
-- this section will be filled in in due course
+- GstRTSPMediaFactory gained API to disable RTCP
+ (gst_rtsp_media_factory_set_enable_rtcp(), "enable-rtcp" property).
+ Previously RTCP was always allowed for all RTSP medias. With this
+ change it is possible to disable RTCP completely, no matter if the
+ client wants to do RTCP or not.
+
+- Make a mount point of / work correctly. While not allowed by the
+ RTSP 2 spec, the RTSP 1 spec is silent on this and it is used in the
+ wild. It is now possible to use / as a mount path in
+ gst-rtsp-server, e.g. rtsp://example.com/ would work with this now.
+ Note that query/fragment parts of the URI are not necessarily
+ correctly handled, and behaviour will differ between various
+ client/server implementations; so use it if you must but don’t bug
+ us if it doesn’t work with third party clients as you’d hoped.
+
+- multithreading fixes (races, refcounting issues, deadlocks)
+
+- ONVIF audio backchannel fixes
+
+- ONVIF trick mode optimisations
+
+- rtspclientsink: new "update-sdp" signal that allows updating the SDP
+ before sending it to the server via ANNOUNCE. This can be used to
+ add additional metadata to the SDP, for example. The order and
+ number of medias must not be changed, however.
GStreamer VAAPI
-- this section will be filled in in due course
+- new AV1 decoder element (vaapiav1dec)
+
+- H264 decoder: handle stereoscopic 3D video with frame packing
+ arrangement SEI messages
+
+- H265 encoder: added Screen Content Coding extensions support
+
+- H265 decoder: gained MAIN_444_12 profile support (decoded to
+ Y412_LE), and 4:2:2 12-bits support (decoded to Y212_LE)
+
+- vaapipostproc: gained BT2020 color standard support
+
+- vaapidecode: now generates caps templates dynamically at runtime in
+ order to advertise actually supported caps instead of all
+ theoretically supported caps.
+
+- GST_VAAPI_DRM_DEVICE environment variable to force a specified DRM
+ device when a DRM display is used. It is ignored when other types of
+ displays are used. By default /dev/dri/renderD128 is used for DRM
+ display.
GStreamer OMX
-- this section will be filled in in due course
+- subframe support in H.264/H.265 decoders
GStreamer Editing Services and NLE
-- this section will be filled in in due course
+- framepositioner: new "operator" property to access blending modes in
+ the compositor
+- timeline: Implement snapping to markers
+- smart-mixer: Add support for d3d11compositor and glvideomixer
+- titleclip: add "draw-shadow" child property
+- ges:// URI support to define a timeline from a description.
+- command-line-formatter
+ - Add track management to timeline description
+ - Add keyframe support
+- ges-launch-1.0:
+ - Add an interactive mode where we can seek etc…
+ - Add option to forward tags
+ - Allow using an existing clip to determine the rendering format
+ (both topology and profile) via new --profile-from command line
+ argument.
+- Fix static build
GStreamer validate
-- this section will be filled in in due course
+- report: Add a way to force backtraces on reports even if not a
+ critical issue (GST_VALIDATE_ISSUE_FLAGS_FORCE_BACKTRACE)
+- Add a flag to gst_validate_replace_variables_in_string() allow
+ defining how to resolve variables in structs
+- Add gst_validate_bin_monitor_get_scenario() to get the bin monitor
+ scenario, which is useful for applications that use Validate
+ directly.
+- Add an expected-values parameter to wait, message-type=XX allowing
+ more precise filtering of the message we are waiting for.
+- Add config file support: each test can now use a config file for the
+ given media file used to test.
+- Add support to check properties of object properties
+- scenario: Add an "action-done" signal to signal when an action is
+ done
+- scenario: Add a "run-command" action type
+- scenario: Allow forcing running action on idle from scenario file
+- scenario: Allow iterating over arrays in foreach
+- scenario: Rename ‘interlaced’ action to ‘non-blocking’
+- scenario: Add a non-blocking flag to the wait signal
GStreamer Python Bindings
-- this section will be filled in in due course
+- Fixes for Python 3.10
+- Various build fixes
+- at least one known breaking change caused by g-i annotation changes
+ (see below)
GStreamer C# Bindings
-- this section will be filled in in due course
+- Fix GstDebugGraphDetails enum
+- Updated to latests GtkSharp
+- Updated to include GStreamer 1.20 API
GStreamer Rust Bindings and Rust Plugins
-The GStreamer Rust bindings are released separately with a different
-release cadence that’s tied to gtk-rs, but the latest release has
-already been updated for the upcoming new GStreamer 1.20 API.
-
-gst-plugins-rs, the module containing GStreamer plugins written in Rust,
-has also seen lots of activity with many new elements and plugins.
-
-What follows is a list of elements and plugins available in
-gst-plugins-rs, so people don’t miss out on all those potentially useful
-elements that have no C equivalent.
-
-- FIXME: add new elements
-
-Rust audio plugins
-
-- audiornnoise: New element for audio denoising which implements the
- noise removal algorithm of the Xiph RNNoise library, in Rust
-- rsaudioecho: Port of the audioecho element from gst-plugins-good
- rsaudioloudnorm: Live audio loudness normalization element based on
- the FFmpeg af_loudnorm filter
-- claxondec: FLAC lossless audio codec decoder element based on the
- pure-Rust claxon implementation
-- csoundfilter: Audio filter that can use any filter defined via the
- Csound audio programming language
-- lewtondec: Vorbis audio decoder element based on the pure-Rust
- lewton implementation
-
-Rust video plugins
-
-- cdgdec/cdgparse: Decoder and parser for the CD+G video codec based
- on a pure-Rust CD+G implementation, used for example by karaoke CDs
-- cea608overlay: CEA-608 Closed Captions overlay element
-- cea608tott: CEA-608 Closed Captions to timed-text (e.g. VTT or SRT
- subtitles) converter
-- tttocea608: CEA-608 Closed Captions from timed-text converter
-- mccenc/mccparse: MacCaption Closed Caption format encoder and parser
-- sccenc/sccparse: Scenarist Closed Caption format encoder and parser
-- dav1dec: AV1 video decoder based on the dav1d decoder implementation
- by the VLC project
-- rav1enc: AV1 video encoder based on the fast and pure-Rust rav1e
- encoder implementation
-- rsflvdemux: Alternative to the flvdemux FLV demuxer element from
- gst-plugins-good, not feature-equivalent yet
-- rsgifenc/rspngenc: GIF/PNG encoder elements based on the pure-Rust
- implementations by the image-rs project
-
-Rust text plugins
-
-- textwrap: Element for line-wrapping timed text (e.g. subtitles) for
- better screen-fitting, including hyphenation support for some
- languages
-
-Rust network plugins
-
-- reqwesthttpsrc: HTTP(S) source element based on the Rust
- reqwest/hyper HTTP implementations and almost feature-equivalent
- with the main GStreamer HTTP source souphttpsrc
-- s3src/s3sink: Source/sink element for the Amazon S3 cloud storage
-- awstranscriber: Live audio to timed text transcription element using
- the Amazon AWS Transcribe API
-
-Generic Rust plugins
-
-- sodiumencrypter/sodiumdecrypter: Encryption/decryption element based
- on libsodium/NaCl
-- togglerecord: Recording element that allows to pause/resume
- recordings easily and considers keyframe boundaries
-- fallbackswitch/fallbacksrc: Elements for handling potentially
- failing (network) sources, restarting them on errors/timeout and
- showing a fallback stream instead
-- threadshare: Set of elements that provide alternatives for various
- existing GStreamer elements but allow to share the streaming threads
- between each other to reduce the number of threads
-- rsfilesrc/rsfilesink: File source/sink elements as replacements for
- the existing filesrc/filesink elements
+- The GStreamer Rust bindings are released separately with a different
+ release cadence that’s tied to gtk-rs, but the latest release has
+ already been updated for the upcoming new GStreamer 1.20 API (v1_20
+ feature).
+
+- gst-plugins-rs, the module containing GStreamer plugins written in
+ Rust, has also seen lots of activity with many new elements and
+ plugins. See the New Elements section above for a list of new Rust
+ elements.
Build and Dependencies
- Meson 0.59 or newer is required to build GStreamer now.
-- FIXME: this section will be filled in in due course
+- The GLib requirement has been bumped to GLib 2.56 or newer (from
+ March 2018).
+
+- The wpe plugin now requires wpe >= 2.28 and wpebackend-fdo >= 1.8
Explicit opt-in required for build of certain plugins with (A)GPL dependencies
See Building plugins with (A)GPL-licensed dependencies for more details
and a non-exhaustive list of plugins affected.
-gst-build: replaced by Monorepo
+gst-build: replaced by mono repository
-- this section will be filled in in due course
-
-- FIXME: describe + link to Monorepo FAQ
+See mono repository section above and the GStreamer mono repository FAQ.
Cerbero
on platforms where dependencies are not readily available, such as
Windows, Android, iOS and macOS.
-General improvements
+General Cerbero improvements
-- this section will be filled in in due course
+- Plugin removed: libvisual
+- New plugins: rtpmanagerbad and rist
-macOS / iOS
+macOS / iOS specific Cerbero improvements
-- this section will be filled in in due course
+- XCode 12 support
+- macOS OS release support is now future-proof, similar to iOS
+- macOS Apple Silicon (ARM64) cross-compile support has been added
+- macOS Apple Silicon (ARM64) native support is currently experimental
-Windows
+Windows specific Cerbero improvements
-- this section will be filled in in due course
+- Visual Studio 2022 support has been added
+- bootstrap is faster since it requires building fewer build-tools
+ recipes on Windows
+- package is faster due to better scheduling of recipe stages and
+ elimination of unnecessary autotools regeneration
+- The following plugins are no longer built on Windows:
+ - a52dec (another decoder is still available in libav)
+ - dvdread
+ - resindvd
Windows MSI installer
-- this section will be filled in in due course
+- no major changes
-Linux
+Linux specific Cerbero improvements
-- this section will be filled in in due course
+- Fedora, Debian OS release support is now more future-proof
+- Amazon Linux 2 support has been added
-Android
+Android specific Cerbero improvements
-- this section will be filled in in due course
+- no major changes
Platform-specific changes and improvements
Android
-- this section will be filled in in due course
+- No major changes
macOS and iOS
-- this section will be filled in in due course
+- applemedia: add ProRes support to vtenc and vtdec
Windows
-- this section will be filled in in due course
+- On Windows the high-resolution clock is enabled now in the
+ gst-launch-1.0 and gst-play-1.0 command line tools, which provides
+ better clock and timer performance on Windows, at the cost of higher
+ power consumption. By default, without the high-resolution clock
+ enabled, the timer precision on Windows is system-dependent and may
+ be as bad as 15ms which is not good enough for many multimedia
+ applications. Developers may want to do the same in their Windows
+ applications if they think it’s a good idea for their application
+ use case, and depending on the Windows version they target. This is
+ not done automatically by GStreamer because on older Windows
+ versions (pre-Windows 10) this affects a global Windows setting and
+ also there’s a power consumption vs. performance trade-off that may
+ differ from application to application.
+
+- dxgiscreencapsrc now supports resolution changes
+
+- The wasapi2 audio plugin was rewritten and now has a higher rank
+ than the old wasapi plugin since it has a number of additional
+ features such as automatic stream routing, and no
+ known-but-hard-to-fix issues. The plugin is always built if the
+ Windows 10 SDK is available now.
+
+- The wasapi device providers now detect and notify dynamic device
+ additions/removals
+
+- d3d11screencapturesrc: new desktop capture element, including
+ GstDeviceProvider implementation to enumerate/select target monitors
+ for capture.
+
+- Direct3D11/DXVA decoder now supports AV1 and MPEG2 codecs
+ (d3d11av1dec, d3d11mpeg2dec)
+
+- VP9 decoding got more reliable and stable thanks to a newly written
+ codec parser
+
+- Support for decoding interlaced H.264/AVC streams
+
+- Hardware-accelerated video deinterlacing (d3d11deinterlace) and
+ video mixing (d3d11compositor)
+
+- Video mixing with the Direct3D11 API (d3d11compositor)
+
+- MediaFoundation API based hardware encoders gained the ability to
+ receive Direct3D11 textures as an input
+
+- Seungha’s blog post “GStreamer ❤ Windows: A primer on the cool stuff
+ you’ll find in the 1.20 release” describes many of the
+ Windows-related improvements in more detail
Linux
-- this section will be filled in in due course
+- bluez: LDAC Bluetooth audio codec support in a2dpsink and avdtpsink,
+ as well as an LDAC RTP payloader (rtpldacpay) and an LDAC audio
+ encoder (ldacenc)
+
+- kmssink: gained support for NV24, NV61, RGB16/BGR16 formats;
+ auto-detect NVIDIA Tegra driver
Documentation improvements
-- this section will be filled in in due course
+- hardware-accelerated GPU plugins will now no longer always list all
+ the element variants for all available GPUs, since those are
+ system-dependent and it’s confusing for users to see those in the
+ documentation just because the GStreamer developer who generated the
+ docs had multiple GPUs to play with at the time. Instead just show
+ the default elements.
-Possibly Breaking Changes
+Possibly Breaking and Other Noteworthy Behavioural Changes
+
+- gst_parse_launch(), gst_parse_bin_from_description() and friends
+ will now error out when setting properties that don’t exist on
+ top-level bins. They were silently ignored before.
+
+- The GstWebRTC library does not expose any objects anymore with
+ public fields. Instead properties have been added to replace that
+ functionality. If you are accessing such fields in your application,
+ switch to the corresponding properties.
-- FIXME: this section will be filled in in due course
-- MPEG-TS SCTE-35 API changes (FIXME: flesh out)
-- gst_parse_launch() and friends now error out on non-existing
- properties on top-level bins where they would silently fail and
- ignore those before.
- playbin and uridecodebin now emit the source-setup signal before the
element is added to the bin and linked so that the source element is
already configured before any scheduling query comes in, which is
- useful for elements such as appsrc or giostreamsrc. (Merge Request)
+ useful for elements such as appsrc or giostreamsrc.
+
+- The source element inside urisourcebin (used inside uridecodebin3
+ which is used inside playbin3) is no longer called "source". This
+ shouldn’t affect anyone hopefully, because there’s a "setup-source"
+ signal to configure the source element and no one should rely on
+ names of internal elements anyway.
+
+- The vp8enc element now expects bps (bits per second) for the
+ "temporal-scalability-target-bitrate" property, which is consistent
+ with the "target-bitrate" property. Since additional configuration
+ is required with modern libvpx to make temporal scaling work anyway,
+ chances are that very few people will have been using this property
+
+- vp8enc and vp9enc now default to “good quality” for the "deadline"
+ property rather then “best quality”. Having the deadline set to best
+ quality causes the encoder to be absurdly slow, most real-life users
+ will want the good quality tradeoff instead.
+
+- The experimental GstTranscoder library API in gst-plugins-bad was
+ changed from a GObject signal-based notification mechanism to a
+ GstBus/message-based mechanism akin to GstPlayer/GstPlay.
+
+- MPEG-TS SCTE-35 API: semantic change for SCTE-35 splice commands:
+ timestamps passed by the application should be in running time now,
+ since users of the API can’t really be expected to predict the local
+ PTS of the muxer.
+
+- The GstContext used by souphttpsrc to share the session between
+ multiple element instances has changed. Previously it provided
+ direct access to the internal SoupSession object, now it only
+ provides access to an opaque, internal type. This change is
+ necessary because SoupSession is not thread-safe at all and can’t be
+ shared safely between arbitrary external code and souphttpsrc.
+
+- Python bindings: GObject-introspection related Annotation fixes have
+ led to a case of a GstVideo.VideoInfo-related function signature
+ changing in the Python bindings (possibly one or two other cases
+ too). This is for a function that should never have been exposed in
+ the first place though, so the bindings are being updated to throw
+ an exception in that case, and the correct replacement API has been
+ added in form of an override.
Known Issues
-- this section will be filled in in due course
-
-- There are a couple of known WebRTC-related regressions/blockers:
-
- - webrtc: DTLS setup with Chrome is broken
- - webrtcbin: First keyframe is usually lost
+- nothing in particular at this point (but also see possibly breaking
+ changes section above)
Contributors
-- this section will be filled in in due course
+Aaron Boxer, Adam Leppky, Adam Williamson, Alba Mendez, Alejandro
+González, Aleksandr Slobodeniuk, Alexander Vandenbulcke, Alex Ashley,
+Alicia Boya García, Andika Triwidada, Andoni Morales Alastruey, Andrew
+Wesie, Andrey Moiseev, Antonio Ospite, Antonio Rojas, Arthur Crippa
+Búrigo, Arun Raghavan, Ashley Brighthope, Axel Kellermann, Baek, Bastien
+Nocera, Bastien Reboulet, Benjamin Gaignard, Bing Song, Binh Truong,
+Biswapriyo Nath, Brad Hards, Brad Smith, Brady J. Garvin, Branko
+Subasic, Camilo Celis Guzman, Chris Bass, ChrisDuncanAnyvision, Chris
+White, Corentin Damman, Daniel Almeida, Daniel Knobe, Daniel Stone,
+david, David Fernandez, David Keijser, David Phung, Devarsh Thakkar,
+Dinesh Manajipet, Dmitry Samoylov, Dmitry Shusharin, Dominique Martinet,
+Doug Nazar, Ederson de Souza, Edward Hervey, Emmanuel Gil Peyrot,
+Enrique Ocaña González, Ezequiel Garcia, Fabian Orccon, Fabrice
+Fontaine, Fernando Jimenez Moreno, Florian Karydes, Francisco Javier
+Velázquez-García, François Laignel, Frederich Munch, Fredrik Pålsson,
+George Kiagiadakis, Georg Lippitsch, Göran Jönsson, Guido Günther,
+Guillaume Desmottes, Guiqin Zou, Haakon Sporsheim, Haelwenn (lanodan)
+Monnier, Haihao Xiang, Haihua Hu, Havard Graff, He Junyan, Helmut
+Januschka, Henry Wilkes, Hosang Lee, Hou Qi, Ignacio Casal Quinteiro,
+Igor Kovalenko, Ilya Kreymer, Imanol Fernandez, Jacek Tomaszewski, Jade
+Macho, Jakub Adam, Jakub Janků, Jan Alexander Steffens (heftig), Jan
+Schmidt, Jason Carrete, Jason Pereira, Jay Douglass, Jeongki Kim, Jérôme
+Laheurte, Jimmi Holst Christensen, Johan Sternerup, John Hassell, John
+Lindgren, John-Mark Bell, Jonathan Matthew, Jordan Petridis, Jose
+Quaresma, Julian Bouzas, Julien, Kai Uwe Broulik, Kasper Steensig
+Jensen, Kellermann Axel, Kevin Song, Khem Raj, Knut Inge Hvidsten, Knut
+Saastad, Kristofer Björkström, Lars Lundqvist, Lawrence Troup, Lim Siew
+Hoon, Lucas Stach, Ludvig Rappe, Luis Paulo Fernandes de Barros, Luke
+Yelavich, Mads Buvik Sandvei, Marc Leeman, Marco Felsch, Marek Vasut,
+Marian Cichy, Marijn Suijten, Marius Vlad, Markus Ebner, Mart Raudsepp,
+Matej Knopp, Mathieu Duponchelle, Matthew Waters, Matthieu De Beule,
+Mengkejiergeli Ba, Michael de Gans, Michael Olbrich, Michael Tretter,
+Michal Dzik, Miguel Paris, Mikhail Fludkov, mkba, Nazar Mokrynskyi,
+Nicholas Jackson, Nicola Murino, Nicolas Dufresne, Niklas Hambüchen,
+Nikolay Sivov, Nirbheek Chauhan, Olivier Blin, Olivier Crete, Olivier
+Crête, Paul Goulpié, Per Förlin, Peter Boba, P H, Philippe Normand,
+Philipp Zabel, Pieter Willem Jordaan, Piotrek Brzeziński, Rafał
+Dzięgiel, Rafostar, raghavendra, Raghavendra, Raju Babannavar, Raleigh
+Littles III, Randy Li, Randy Li (ayaka), Ratchanan Srirattanamet, Raul
+Tambre, reed.lawrence, Ricky Tang, Robert Rosengren, Robert Swain, Robin
+Burchell, Roman Sivriver, R S Nikhil Krishna, Ruben Gonzalez, Ruslan
+Khamidullin, Sanchayan Maity, Scott Moreau, Sebastian Dröge, Sergei
+Kovalev, Seungha Yang, Sid Sethupathi, sohwan.park, Sonny Piers, Staz M,
+Stefan Brüns, Stéphane Cerveau, Stephan Hesse, Stian Selnes, Stirling
+Westrup, Théo MAILLART, Thibault Saunier, Tim, Timo Wischer, Tim-Philipp
+Müller, Tim Schneider, Tobias Ronge, Tom Schoonjans, Tulio Beloqui,
+tyler-aicradle, U. Artie Eoff, Ung, Val Doroshchuk, VaL Doroshchuk,
+Víctor Manuel Jáquez Leal, Vivek R, Vivia Nikolaidou, Vivienne
+Watermeier, Vladimir Menshakov, Will Miller, Wim Taymans, Xabier
+Rodriguez Calvar, Xavier Claessens, Xℹ Ruoyao, Yacine Bandou, Yinhang
+Liu, youngh.lee, youngsoo.lee, yychao, Zebediah Figura, Zhang yuankun,
+Zhang Yuankun, Zhao, Zhao Zhili, , Aleksandar Topic, Antonio Ospite,
+Bastien Nocera, Benjamin Gaignard, Brad Hards, Carlos Falgueras García,
+Célestin Marot, Corentin Damman, Corentin Noël, Daniel Almeida, Daniel
+Knobe, Danny Smith, Dave Piché, Dmitry Osipenko, Fabrice Fontaine,
+fjmax, Florian Zwoch, Guillaume Desmottes, Haihua Hu, Heinrich Kruger,
+He Junyan, Jakub Adam, James Cowgill, Jan Alexander Steffens (heftig),
+Jean Felder, Jeongki Kim, Jiri Uncovsky, Joe Todd, Jordan Petridis,
+Krystian Wojtas, Marc-André Lureau, Marcin Kolny, Marc Leeman, Mark
+Nauwelaerts, Martin Reboredo, Mathieu Duponchelle, Matthew Waters,
+Mengkejiergeli Ba, Michael Gruner, Nicolas Dufresne, Nirbheek Chauhan,
+Olivier Crête, Philippe Normand, Rafał Dzięgiel, Ralf Sippl, Robert
+Mader, Sanchayan Maity, Sangchul Lee, Sebastian Dröge, Seungha Yang,
+Stéphane Cerveau, Teh Yule Kim, Thibault Saunier, Thomas Klausner, Timo
+Wischer, Tim-Philipp Müller, Tobias Reineke, Tomasz Andrzejak, Trung Do,
+Tyler Compton, Ung, Víctor Manuel Jáquez Leal, Vivia Nikolaidou, Wim
+Taymans, wngecn, Wonchul Lee, wuchang li, Xavier Claessens, Xi Ruoyao,
+Yoshiharu Hirose, Zhao,
… and many others who have contributed bug reports, translations, sent
suggestions or helped testing.
1.20.0
-1.20.0 is scheduled to be released around October/November 2021.
+1.20.0 is scheduled to be released around early February 2022.
Schedule for 1.22
unstable development version leading up to the stable 1.22 release. The
development of 1.21/1.22 will happen in the git main branch.
-The plan for the 1.22 development cycle is yet to be confirmed.
+The plan for the 1.22 development cycle is yet to be confirmed. Assuming
+no major project-wide reorganisations in the 1.22 cycle we might try and
+aim for a release around August 2022.
1.22 will be backwards-compatible to the stable 1.20, 1.18, 1.16, 1.14,
1.12, 1.10, 1.8, 1.6, 1.4, 1.2 and 1.0 release series.
------------------------------------------------------------------------
These release notes have been prepared by Tim-Philipp Müller with
-contributions from …
+contributions from Matthew Waters, Nicolas Dufresne, Nirbheek Chauhan,
+Sebastian Dröge and Seungha Yang.
License: CC BY-SA 4.0
-This is GStreamer gst-plugins-bad 1.19.3.
+This is GStreamer gst-plugins-bad 1.19.90.
GStreamer 1.19 is the development branch leading up to the next major
stable version which will be 1.20.
<release>
<Version>
+ <revision>1.19.90</revision>
+ <branch>main</branch>
+ <name></name>
+ <created>2022-01-28</created>
+ <file-release rdf:resource="https://gstreamer.freedesktop.org/src/gst-plugins-bad/gst-plugins-bad-1.19.90.tar.xz" />
+ </Version>
+ </release>
+
+ <release>
+ <Version>
<revision>1.19.3</revision>
<branch>main</branch>
<name></name>
project('gst-plugins-bad', 'c', 'cpp',
- version : '1.19.3.1',
+ version : '1.19.90',
meson_version : '>= 0.59',
default_options : [ 'warning_level=1',
'buildtype=debugoptimized' ])
+=== release 1.19.90 ===
+
+2022-01-28 14:28:35 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * NEWS:
+ * RELEASE:
+ * gst-plugins-base.doap:
+ * meson.build:
+ Release 1.19.90
+
+2022-01-28 14:28:28 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * ChangeLog:
+ Update ChangeLogs for 1.19.90
+
2022-01-27 14:22:26 +0200 Sebastian Dröge <sebastian@centricular.com>
* gst-libs/gst/rtp/gstrtphdrext.c:
GStreamer 1.20 Release Notes
GStreamer 1.20 has not been released yet. It is scheduled for release in
-November 2021.
+late January / early February 2022.
1.19.x is the unstable development version that is being developed in
-the git main branch and which will eventually result in 1.20, and 1.19.3
-is the current development release in that series
-
-It is expected that feature freeze will be in early October 2021,
-followed by one or two 1.19.9x pre-releases and the new 1.20 stable
-release around October/November 2021.
+the git main branch and which will eventually result in 1.20, and
+1.19.90 is the first release candidate in that series (1.20rc1).
1.20 will be backwards-compatible to the stable 1.18, 1.16, 1.14, 1.12,
-1.10, 1.8, 1.6,, 1.4, 1.2 and 1.0 release series.
+1.10, 1.8, 1.6, 1.4, 1.2 and 1.0 release series.
See https://gstreamer.freedesktop.org/releases/1.20/ for the latest
version of this document.
-Last updated: Monday 1 November 2021, 01:00 UTC (log)
+Last updated: Wednesday 26 January 2022, 01:00 UTC (log)
Introduction
Highlights
-- this section will be completed in due course
+- Development in GitLab was switched to a single git repository
+ containing all the modules
+- GstPlay: new high-level playback library, replaces GstPlayer
+- WebM Alpha decoding support
+- Encoding profiles can now be tweaked with additional
+ application-specified element properties
+- Compositor: multi-threaded video conversion and mixing
+- RTP header extensions: unified support in RTP depayloader and
+ payloader base classes
+- SMPTE 2022-1 2-D Forward Error Correction support
+- Smart encoding (passthrough) support for VP8, VP9, H.265 in
+ encodebin and transcodebin
+- Runtime compatibility support for libsoup2 and libsoup3 (libsoup3
+ support experimental)
+- Video decoder subframe support
+- Video decoder automatic packet-loss, data corruption, and keyframe
+ request handling for RTP / WebRTC / RTSP
+- MP4 and Matroska muxers now support profile/level/resolution changes
+ for H264/H265 input streams (i.e. codec data changing on the fly)
+- MP4 muxing mode that initially creates a fragmented mp4 which is
+ converted to a regular mp4 on EOS
+- Audio support for the WebKit Port for Embedded (WPE) web page source
+ element
+- CUDA based video color space convert and rescale elements and
+ upload/download elements
+- NVIDIA memory:NVMM support for OpenGL glupload and gldownload
+ elements
+- Many WebRTC improvements
+- The new VA-API plugin implemention fleshed out with more decoders
+ and new postproc elements
+- AppSink API to retrieve events in addition to buffers and buffer
+ lists
+- AppSrc gained more configuration options for the internal queue
+ (leakiness, limits in buffers and time, getters to read current
+ levels)
+- Updated Rust bindings and many new Rust plugins
+- Improved support for custom minimal GStreamer builds
+- Support build against FFmpeg 5.0
+- Linux Stateless CODEC support gained MPEG2 and VP9
+- Windows Direct3D11/DXVA decoder gained AV1 and MPEG2 support
+- Lots of new plugins, features, performance improvements and bug
+ fixes
Major new features and changes
Noteworthy new features and API
-- this section will be filled in in due course
+- gst_element_get_request_pad() has been deprecated in favour of the
+ newly-added gst_element_request_pad_simple() which does the exact
+ same thing but has a less confusing name that hopefully makes clear
+ that the function request a new pad rather than just retrieves an
+ already-existing request pad.
+
+Development in GitLab was switched to a single git repository containing all the modules
+
+The GStreamer multimedia framework is a set of libraries and plugins
+split into a number of distinct modules which are released independently
+and which have so far been developed in separate git repositories in
+freedesktop.org GitLab.
+
+In addition to these separate git repositories there was a gst-build
+module that would use the Meson build systems’s subproject feature to
+download each individual module and then build everything in one go. It
+would also provide an uninstalled development environment that made it
+easy to work on GStreamer and use or test versions other than the
+system-installed GStreamer version.
+
+All of these modules have now (as of 28 September 2021) been merged into
+a single git repository (“Mono repository” or “monorepo”) which should
+simplify development workflows and continuous integration, especially
+where changes need to be made to multiple modules at once.
+
+This mono repository merge will primarily affect GStreamer developers
+and contributors and anyone who has workflows based on the GStreamer git
+repositories.
+
+The Rust bindings and Rust plugins modules have not been merged into the
+mono repository at this time because they follow a different release
+cycle.
+
+The mono repository lives in the existing GStreamer core git repository
+in GitLab in the new main branch and all future development will happen
+on this branch.
+
+Modules will continue to be released as separate tarballs.
+
+For more details, please see the GStreamer mono repository FAQ.
+
+GstPlay: new high-level playback library replacing GstPlayer
+
+- GstPlay is a new high-level playback library that replaces the older
+ GstPlayer API. It is basically the same API as GstPlayer but
+ refactored to use bus messages for application notifications instead
+ of GObject signals. There is still a signal adapter object for those
+ who prefer signals. Since the existing GstPlayer API is already in
+ use in various applications, it didn’t seem like a good idea to
+ break it entirely. Instead a new API was added, and it is expected
+ that this new GstPlay API will be moved to gst-plugins-base in
+ future.
+
+- The existing GstPlayer API is scheduled for deprecation and will be
+ removed at some point in the future (e.g. in GStreamer 1.24), so
+ application developers are urged to migrate to the new GstPlay API
+ at their earliest convenience.
+
+WebM alpha decoding
+
+- Implement WebM alpha decoding (VP8/VP9 with alpha), which required
+ support and additions in various places. This is supported both with
+ software decoders and hardware-accelerated decoders.
+
+- VP8/VP9 don’t support alpha components natively in the codec, so the
+ way this is implemented in WebM is by encoding the alpha plane with
+ transparency data as a separate VP8/VP9 stream. Inside the WebM
+ container (a variant of Matroska) this is coded as a single video
+ track with the “normal” VP8/VP9 video data making up the main video
+ data and each frame of video having an encoded alpha frame attached
+ to it as extra data ("BlockAdditional").
+
+- matroskademux has been extended extract this per-frame alpha side
+ data and attach it in form of a GstVideoCodecAlphaMeta to the
+ regular video buffers. Note that this new meta is specific to this
+ VP8/VP9 alpha support and can’t be used to just add alpha support to
+ other codecs that don’t support it. Lastly, matroskademux also
+ advertises the fact that the streams contain alpha in the caps.
+
+- The new codecalpha plugin contains various bits of infrastructure to
+ support autoplugging and debugging:
+
+ - codecalphademux splits out the alpha stream from the metas on
+ the regular VP8/VP9 buffers
+ - alphacombine takes two decoded raw video streams (one alpha, one
+ the regular video) and combines it into a video stream with
+ alpha
+ - vp8alphadecodebin + vp9alphadecodebin are wrapper bins that use
+ the regular vp8dec and vp9dec software decoders to decode
+ regular and alpha streams and combine them again. To decodebin
+ these look like regular decoders which ju
+ - The V4L2 CODEC plugin has stateless VP8/VP9 decoders that can
+ decode both alpha and non-alpha stream with a single decoder
+ instance
+
+- A new AV12 video format was added which is basically NV12 with an
+ alpha plane, which is more convenient for many hardware-accelerated
+ decoders.
+
+- Watch Nicolas Dufresne’s LCA 2022 talk “Bringing WebM Alpha support
+ to GStreamer” for all the details and a demo.
+
+RTP Header Extensions Base Class and Automatic Header Extension Handling in RTP Payloaders and Depayloaders
+
+- RTP Header Extensions are specified in RFC 5285 and provide a way to
+ add small pieces of data to RTP packets in between the RTP header
+ and the RTP payload. This is often used for per-frame metadata,
+ extended timestamps or other application-specific extra data. There
+ are several commonly-used extensions specified in various RFCs, but
+ senders are free to put any kind of data in there, as long as sender
+ and receiver both know what that data is. Receivers that don’t know
+ about the header extensions will just skip the extra data without
+ ever looking at it. These header extensions can often be combined
+ with any kind of payload format, so may need to be supported by many
+ RTP payloader and depayloader elements.
+
+- Inserting and extracting RTP header extension data has so far been a
+ bit inconvenient in GStreamer: There are functions to add and
+ retrieve RTP header extension data from RTP packets, but nothing
+ works automatically, even for common extensions. People would have
+ to do the insertion/extraction either in custom elements
+ before/after the RTP payloader/depayloader, or inside pad probes,
+ which isn’t very nice.
+
+- This release adds various pieces of new infrastructure for generic
+ RTP header extension handling, as well as some implementations for
+ common extensions:
+
+ - GstRTPHeaderExtension is a new helper base class for reading and
+ writing RTP header extensions. Nominally this subclasses
+ GstElement, but only so these extensions are stored in the
+ registry where they can be looked up by URI or name. They don’t
+ have pads and don’t get added to the pipeline graph as an
+ element.
+
+ - "add-extension" and "clear-extension" action signals on RTP
+ payloaders and depayloaders for manual extension management
+
+ - The "request-extension" signal will be emitted if an extension
+ is encountered that requires explicit mapping by the application
+
+ - new "auto-header-extension" property on RTP payloaders and
+ depayloaders for automatic handling of known header extensions.
+ This is enabled by default. The extensions must be signalled via
+ caps / SDP.
+
+ - RTP header extension implementations:
+
+ - rtphdrextclientaudiolevel: Client-to-Mixer Audio Level
+ Indication (RFC 6464) (also see below)
+ - rtphdrextcolorspace: Color Space extension, extends RTP
+ packets with color space and high dynamic range (HDR)
+ information
+ - rtphdrexttwcc: Transport Wide Congestion Control support
+
+- gst_rtp_buffer_remove_extension_data() is a new helper function to
+ remove an RTP header extension from an RTP buffer
+
+- The existing gst_rtp_buffer_set_extension_data() now also supports
+ shrinking the extension data in size
+
+AppSink and AppSrc improvements
+
+- appsink: new API to pull events out of appsink in addition to
+ buffers and buffer lists.
+
+ There was previously no way for users to receive incoming events
+ from appsink properly serialised with the data flow, even if they
+ are serialised events. The reason for that is that the only way to
+ intercept events was via a pad probe on the appsink sink pad, but
+ there is also internal queuing inside of appsink, so it’s difficult
+ to ascertain the right order of everything in all cases.
+
+ There is now a new "new-serialized-event" signal which will be
+ emitted when there’s a new event pending (just like the existing
+ "new-sample" signal). The "emit-signals" property must be set to
+ TRUE in order to activate this (but it’s also fine to just pull from
+ the application thread without using the signals).
+
+ gst_app_sink_pull_object() and gst_app_sink_try_pull_object() can be
+ used to pull out either an event or a new sample carrying a buffer
+ or buffer list, whatever is next in the queue.
+
+ EOS events will be filtered and will not be returned. EOS handling
+ can be done the ususal way, same as with _pull_sample().
+
+- appsrc: allow configuration of internal queue limits in time and
+ buffers and add leaky mode.
+
+ There is internal queuing inside appsrc so the application thread
+ can push data into the element which will then be picked up by the
+ source element’s streaming thread and pushed into the pipeline from
+ that streaming thread. This queue is unlimited by default and until
+ now it was only possible to set a maximum size limit in bytes. When
+ that byte limit is reached, the pushing thread (application thread)
+ would be blocked until more space becomes available.
+
+ A limit in bytes is not particularly useful for many use cases, so
+ now it is possible to also configure limits in time and buffers
+ using the new "max-time" and "max-buffers" properties. Of course
+ there are also matching new read-only"current-level-buffers" and
+ "current-level-time properties" properties to query the current fill
+ level of the internal queue in time and buffers.
+
+ And as if that wasn’t enough the internal queue can also be
+ configured as leaky using the new "leaky-type" property. That way
+ when the queue is full the application thread won’t be blocked when
+ it tries to push in more data, but instead either the new buffer
+ will be dropped or the oldest data in the queue will be dropped.
+
+Better string serialization of nested GstCaps and GstStructures
+
+- New string serialisation format for structs and caps that can handle
+ nested structs and caps properly by using brackets to delimit nested
+ items (e.g. some-struct, some-field=[nested-struct, nested=true]).
+ Unlike the default format the new variant can also support more than
+ one level of nesting. For backwards-compatibility reasons the old
+ format is still output by default when serialising caps and structs
+ using the existing API. The new functions gst_caps_serialize() and
+ gst_structure_serialize() can be used to output strings in the new
+ format.
+
+Convenience API for custom GstMetas
+
+- New convenience API to register and create custom GstMetas:
+ gst_meta_register_custom() and gst_buffer_add_custom_meta(). Such
+ custom meta is backed by a GstStructure and does not require that
+ users of the API expose their GstMeta implementation as public API
+ for other components to make use of it. In addition, it provides a
+ simpler interface by ignoring the impl vs. api distinction that the
+ regular API exposes. This new API is meant to be the meta
+ counterpart to custom events and messages, and to be more convenient
+ than the lower-level API when the absolute best performance isn’t a
+ requirement. The reason it’s less performant than a “proper” meta is
+ that a proper meta is just a C struct in the end whereas this goes
+ through the GstStructure API which has a bit more overhead, which
+ for most scenarios is negligible however. This new API is useful for
+ experimentation or proprietary metas, but also has some limitations:
+ it can only be used if there’s a single producer of these metas;
+ it’s not allowed to register the same custom meta multiple times or
+ from multiple places.
+
+Additional Element Properties on Encoding Profiles
+
+- GstEncodingProfile: The new "element-properties" and
+ gst_encoding_profile_set_element_properties() API allows
+ applications to set additional element properties on encoding
+ profiles to configure muxers and encoders. So far the encoding
+ profile template was the only place where this could be specified,
+ but often what applications want to do is take a ready-made encoding
+ profile shipped by GStreamer or the application and then tweak the
+ settings on top of that, which is now possible with this API. Since
+ applications can’t always know in advance what encoder element will
+ be used in the end, it’s even possible to specify properties on a
+ per-element basis.
+
+ Encoding Profiles are used in the encodebin, transcodebin and
+ camerabin elements and APIs to configure output formats (containers
+ and elementary streams).
+
+Audio Level Indication Meta for RFC 6464
+
+- New GstAudioLevelMeta containing Audio Level Indication as per RFC
+ 6464
+
+- The level element has been updated to add GstAudioLevelMeta on
+ buffers if the "audio-level-meta" property is set to TRUE. This can
+ then in turn be picked up by RTP payloaders to signal the audio
+ level to receivers through RTP header extensions (see above).
+
+- New Client-to-Mixer Audio Level Indication (RFC6464) RTP Header
+ Extension which should be automatically created and used by RTP
+ payloaders and depayloaders if their "auto-header-extension"
+ property is enabled and if the extension is part of the RTP caps.
+
+Automatic packet loss, data corruption and keyframe request handling for video decoders
+
+- The GstVideoDecoder base class has gained various new APIs to
+ automatically handle packet loss and data corruption better by
+ default, especially in RTP, RTSP and WebRTC streaming scenarios, and
+ to give subclasses more control about how they want to handle
+ missing data:
+
+ - Video decoder subclasses can mark output frames as corrupted via
+ the new GST_VIDEO_CODEC_FRAME_FLAG_CORRUPTED flag
+
+ - A new "discard-corrupted-frames" property allows applications to
+ configure decoders so that corrupted frames are directly
+ discarded instead of being forwarded inside the pipeline. This
+ is a replacement for the "output-corrupt" property of the FFmpeg
+ decoders.
+
+ - RTP depayloaders can now signal to decoders that data is missing
+ when sending GAP events for lost packets. GAP events can be sent
+ for various reason in a GStreamer pipeline. Often they are just
+ used to let downstream elements know that there isn’t a buffer
+ available at the moment, so downstream elements can move on
+ instead of waiting for one. They are also sent by RTP
+ depayloaders in the case that packets are missing, however, and
+ so far a decoder was not able to differentiate the two cases.
+ This has been remedied now: GAP events can be decorated with
+ gst_event_set_gap_flags() and GST_GAP_FLAG_MISSING_DATA to let
+ decoders now what happened, and decoders can then use that in
+ some cases to handle missing data better.
+
+ - The GstVideoDecoder::handle_missing_data vfunc was added to
+ inform subclasses about packet loss or missing data and let them
+ handle it in their own way if they like.
+
+ - gst_video_decoder_set_needs_sync_point() lets subclasses signal
+ that they need the stream to start with a sync point. If
+ enabled, the base class will discard all non-sync point frames
+ in the beginning and after a flush and does not pass them to the
+ subclass. Furthermore, if the first frame is not a sync point,
+ the base class will try and request a sync frame from upstream
+ by sending a force-key-unit event (see next items).
+
+ - New "automatic-request-sync-points" and
+ "automatic-request-sync-point-flags" properties to automatically
+ request sync points when needed, e.g. on packet loss or if the
+ first frame is not a keyframe. Applications may want to enable
+ this on decoders operating in e.g. RTP/WebRTC/RTSP receiver
+ pipelines.
+
+ - The new "min-force-key-unit-interval" property can be used to
+ ensure there’s a minimal interval between keyframe requests to
+ upstream (and/or the sender) and we’re not flooding the sender
+ with key unit requests.
+
+ - gst_video_decoder_request_sync_point() allows subclasses to
+ request a new sync point (e.g. if they choose to do their own
+ missing data handling). This will still honour the
+ "min-force-key-unit-interval" property if set.
+
+Improved support for custom minimal GStreamer builds
+
+- Element registration and registration of other plugin features
+ inside plugin init functions has been improved in order to
+ facilitate minimal custom GStreamer builds.
+
+- A number of new macros have been added to declare and create
+ per-element and per-pluginfeature register functions in all plugins,
+ and then call those from the per-plugin plugin_init functions:
+
+ - GST_ELEMENT_REGISTER_DEFINE,
+ GST_DEVICE_PROVIDER_REGISTER_DEFINE,
+ GST_DYNAMIC_TYPE_REGISTER_DEFINE, GST_TYPE_FIND_REGISTER_DEFINE
+ for the actual registration call with GStreamer
+ - GST_ELEMENT_REGISTER, GST_DEVICE_PROVIDER_REGISTER,
+ GST_DYNAMIC_TYPE_REGISTER, GST_PLUGIN_STATIC_REGISTER,
+ GST_TYPE_FIND_REGISTER to call the registration function defined
+ by the REGISTER_DEFINE macro
+ - GST_ELEMENT_REGISTER_DECLARE,
+ GST_DEVICE_PROVIDER_REGISTER_DECLARE,
+ GST_DYNAMIC_TYPE_REGISTER_DECLARE,
+ GST_TYPE_FIND_REGISTER_DECLARE to declare the registration
+ function defined by the REGISTER_DEFINE macro
+ - and various variants for advanced use cases.
+
+- This means that applications can call the per-element and
+ per-pluginfeature registration functions for only the elements they
+ need instead of registering plugins as a whole with all kinds of
+ elements that may not be required (e.g. encoder and decoder instead
+ of just decoder). In case of static linking all unused functions and
+ their dependencies would be removed in this case by the linker,
+ which helps minimise binary size for custom builds.
+
+- gst_init() will automatically call a gst_init_static_plugins()
+ function if one exists.
+
+- See the GStreamer static build documentation and Stéphane’s blog
+ post Generate a minimal GStreamer build, tailored to your needs for
+ more details.
New elements
-- this section will be filled in in due course
+- New aesdec and aesenc elements for AES encryption and decryption in
+ a custom format.
+
+- New encodebin2 element with dynamic/sometimes source pads in order
+ to support the option of doing the muxing outside of encodebin,
+ e.g. in combination with a splitmuxsink.
+
+- New fakeaudiosink and videocodectestsink elements for testing and
+ debugging (see below for more details)
+
+- rtpisacpay, rtpisacdepay: new RTP payloader and depayloader for iSAC
+ audio codec
+
+- rtpst2022-1-fecdec, rtpst2022-1-fecenc: new elements providing SMPTE
+ 2022-1 2-D Forward Error Correction. More details in Mathieu’s blog
+ post.
+
+- isac: new plugin wrapping the Internet Speech Audio Codec reference
+ encoder and decoder from the WebRTC project.
+
+- asio: plugin for Steinberg ASIO (Audio Streaming Input/Output) API
+
+- gssrc, gssink: add source and sink for Google Cloud Storage
+
+- onnx: new plugin to apply ONNX neural network models to video
+
+- openaptx: aptX and aptX-HD codecs using libopenaptx (v0.2.0)
+
+- qroverlay, debugqroverlay: new elements that allows overlaying data
+ on top of video in form of a QR code
+
+- cvtracker: new OpenCV-based tracker element
+
+- av1parse, vp9parse: new parsers for AV1 and VP9 video
+
+- va: work on the new VA-API plugin implementation for
+ hardware-accelerated video decoding and encoding has continued at
+ pace, with various new decoders and filters having joined the
+ initial vah264dec:
+
+ - vah265dec: VA-API H.265 decoder
+ - vavp8dec: VA-API VP8 decoder
+ - vavp9dec: VA-API VP9 decoder
+ - vaav1dec: VA-API AV1 decoder
+ - vampeg2dec: VA-API MPEG-2 decoder
+ - vadeinterlace: : VA-API deinterlace filter
+ - vapostproc: : VA-API postproc filter (color conversion,
+ resizing, cropping, color balance, video rotation, skin tone
+ enhancement, denoise, sharpen)
+
+ See Víctor’s blog post “GstVA in GStreamer 1.20” for more details
+ and what’s coming up next.
+
+- vaapiav1dec: new AV1 decoder element (in gstreamer-vaapi)
+
+- msdkav1dec: hardware-accelerated AV1 decoder using the Intel Media
+ SDK / oneVPL
+
+- nvcodec plugin for NVIDIA NVCODEC API for hardware-accelerated video
+ encoding and decoding:
+
+ - cudaconvert, cudascale: new CUDA based video color space convert
+ and rescale elements
+ - cudaupload, cudadownload: new helper elements for memory
+ transfer between CUDA and system memory spaces
+ - nvvp8sldec, nvvp9sldec: new GstCodecs-based VP8/VP9 decoders
+
+- Various new hardware-accelerated elements for Windows:
+
+ - d3d11screencapturesrc: new desktop capture element, including a
+ GstDeviceProvider implementation to enumerate/select target
+ monitors for capture.
+ - d3d11av1dec and d3d11mpeg2dec: AV1 and MPEG-2 decoders
+ - d3d11deinterlace: deinterlacing filter
+ - d3d11compositor: video composing element
+ - see Windows section below for more details
+
+- new Rust plugins:
+
+ - audiornnoise: Removes noise from an audio stream
+ - awstranscribeparse: Parses AWS audio transcripts into timed text
+ buffers
+ - ccdetect: Detects if valid closed captions are present in a
+ closed captions stream
+ - cea608tojson: Converts CEA-608 Closed Captions to a JSON
+ representation
+ - cmafmux: CMAF fragmented MP4 muxer
+ - dashmp4mux: DASH fragmented MP4 muxer
+ - isofmp4mux: ISO fragmented MP4 muxer
+ - ebur128level: EBU R128 Loudness Level Measurement
+ - ffv1dec: FFV1 video decoder
+ - gtk4paintablesink: GTK4 video sink, which provides a
+ GdkPaintable that can be rendered in various widgets
+ - hlssink3: HTTP Live Streaming sink
+ - hrtfrender: Head-Related Transfer Function (HRTF) renderer
+ - hsvdetector: HSV colorspace detector
+ - hsvfilter: HSV colorspace filter
+ - jsongstenc: Wraps buffers containing any valid top-level JSON
+ structures into higher level JSON objects, and outputs those as
+ ndjson
+ - jsongstparse: Parses ndjson as output by jsongstenc
+ - jsontovtt: converts JSON to WebVTT subtitles
+ - regex: Applies regular expression operations on text
+ - roundedcorners: Adds rounded corners to video
+ - spotifyaudiosrc: Spotify source
+ - textahead: Display upcoming text buffers ahead (e.g. for
+ Karaoke)
+ - transcriberbin: passthrough bin that transcribes raw audio to
+ closed captions using awstranscriber and puts the captions as
+ metas onto the video
+ - tttojson: Converts timed text to a JSON representation
+ - uriplaylistbin: Playlist source bin
+ - webpdec-rs: WebP image decoder with animation support
+
+- New plugin codecalpha with elements to assist with WebM Alpha
+ decoding
+
+ - codecalphademux: Split stream with GstVideoCodecAlphaMeta into
+ two streams
+ - alphacombine: Combine two raw video stream (I420 or NV12) as one
+ stream with alpha channel (A420 or AV12)
+ - vp8alphadecodebin: A bin to handle software decoding of VP8 with
+ alpha
+ - vp9alphadecodebin: A bin to handle software decoding of VP9 with
+ alpha
+
+- New hardware accelerated elements for Linux:
+
+ - v4l2slmpeg2dec: Support for Linux Stateless MPEG2 decoders
+ - v4l2slvp9dec: Support for Linux Stateless VP9 decoders
+ - v4l2slvp8alphadecodebin: Support HW accelerated VP8 with alpha
+ layer decoding
+ - v4l2slvp9alphadecodebin: Support HW accelerated VP9 with alpha
+ layer decoding
New element features and additions
-- this section will be filled in in due course
+- assrender: handle more font mime types; better interaction with
+ matroskademux for embedded fonts
+
+- audiobuffersplit: Add support for specifying output buffer size in
+ bytes (not just duration)
+
+- audiolatency: new "samplesperbuffer" property so users can configure
+ the number of samples per buffer. The default value is 240 samples
+ which is equivalent to 5ms latency with a sample rate of 48000,
+ which might be larger than actual buffer size of audio capture
+ device.
+
+- audiomixer, audiointerleave, GstAudioAggregator: now keep a count of
+ samples that are dropped or processed as statistic and can be made
+ to post QoS messages on the bus whenever samples are dropped by
+ setting the "qos-messages" property on input pads.
+
+- audiomixer, compositor: improved handling of new inputs added at
+ runtime. New API was added to the GstAggregator base class to allow
+ subclasses to opt into an aggregation mode where inactive pads are
+ ignored when processing input buffers
+ (gst_aggregator_set_ignore_inactive_pads(),
+ gst_aggregator_pad_is_inactive()). An “inactive pad” in this context
+ is a pad which, in live mode, hasn’t yet received a first buffer,
+ but has been waited on at least once. What would happen usually in
+ this case is that the aggregator would wait for data on this pad
+ every time, up to the maximum configured latency. This would
+ inadvertently push mixer elements in live mode to the configured
+ latency envelope and delay processing when new inputs are added at
+ runtime until these inputs have actually produced data. This is
+ usually undesirable. With this new API, new inputs can be added
+ (requested) and configured and they won’t delay the data processing.
+ Applications can opt into this new behaviour by setting the
+ "ignore-inactive-pads" property on compositor, audiomixer or other
+ GstAudioAggregator-based elements.
+
+- cccombiner: implement “scheduling” of captions. So far cccombiner’s
+ behaviour was essentially that of a funnel: it strictly looked at
+ input timestamps to associate together video and caption buffers.
+ Now it will try to smoothly schedule caption buffers in order to
+ have exactly one per output video buffer. This might involve
+ rewriting input captions, for example when the input is CDP then
+ sequence counters are rewritten, time codes are dropped and
+ potentially re-injected if the input video frame had a time code
+ meta. This can also lead to the input drifting from synchronization,
+ when there isn’t enough padding in the input stream to catch up. In
+ that case the element will start dropping old caption buffers once
+ the number of buffers in its internal queue reaches a certain limit
+ (configurable via the "max-scheduled" property). The new original
+ funnel-like behaviour can be restored by setting the "scheduling"
+ property to FALSE.
+
+- ccconverter: new "cdp-mode" property to specify which sections to
+ include in CDP packets (timecode, CC data, service info). Various
+ software, including ffmpeg’s Decklink support, fails parsing CDP
+ packets that contain anything but CC data in the CDP packets.
+
+- clocksync: new "sync-to-first" property for automatic timestamp
+ offset setup: if set clocksync will set up the "ts-offset" value
+ based on the first buffer and the pipeline’s running time when the
+ first buffer arrived. The newly configured "ts-offset" in this case
+ would be the value that allows outputting the first buffer without
+ waiting on the clock. This is useful for example to feed a non-live
+ input into an already-running pipeline.
+
+- compositor:
+
+ - multi-threaded input conversion and compositing. Set the
+ "max-threads" property to activate this.
+ - new "sizing-policy" property to support display aspect ratio
+ (DAR)-aware scaling. By default the image is scaled to fill the
+ configured destination rectangle without padding and without
+ keeping the aspect ratio. With sizing-policy=keep-aspect-ratio
+ the input image is scaled to fit the destination rectangle
+ specified by GstCompositorPad:{xpos, ypos, width, height}
+ properties preserving the aspect ratio. As a result, the image
+ will be centered in the destination rectangle with padding if
+ necessary.
+ - new "zero-size-is-unscaled" property on input pads. By default
+ pad width=0 or pad height=0 mean that the stream should not be
+ scaled in that dimension. But if the "zero-size-is-unscaled"
+ property is set to FALSE a width or height of 0 is instead
+ interpreted to mean that the input image on that pad should not
+ be composited, which is useful when creating animations where an
+ input image is made smaller and smaller until it disappears.
+ - improved handling of new inputs at runtime via
+ "ignore-inactive-pads"property (see above for details)
+ - allow output format with alpha even if none of the inputs have
+ alpha (also glvideomixer and other GstVideoAggregator
+ subclasses)
+
+- dashsink: add h265 codec support and signals for allowing custom
+ playlist/fragment output
+
+- decodebin3:
+
+ - improved decoder selection, especially for hardware decoders
+ - make input activation “atomic” when adding inputs dynamically
+ - better interleave handling: take into account decoder latency
+ for interleave size
+
+- decklink:
+
+ - Updated DeckLink SDK to 11.2 to support DeckLink 8K Pro
+ - decklinkvideosrc:
+ - More accurate and stable capture timestamps: use the
+ hardware reference clock time when the frame was finished
+ being captured instead of a clock time much further down the
+ road.
+ - Automatically detect widescreen vs. normal NTSC/PAL
+
+- encodebin:
+
+ - add “smart encoding” support for H.265, VP8 and VP9 (i.e. only
+ re-encode where needed and otherwise pass through encoded video
+ as-is).
+ - H264/H265 smart encoding improvements: respect user-specified
+ stream-format, but if not specified default to avc3/hvc1 with
+ in-band SPS/PPS/VPS signalling for more flexibility.
+ - new encodebin2 element with dynamic/sometimes source pads in
+ order to support the option of doing the muxing outside of
+ encodebin, e.g. in combination with splitmuxsink.
+ - add APIs to set element properties on encoding profiles (see
+ below)
+
+- errorignore: new "ignore-eos" property to also ignore FLOW_EOS from
+ downstream elements
+
+- giosrc: add support for growing source files: applications can
+ specify that the underlying file being read is growing by setting
+ the "is-growing" property. If set, the source won’t EOS when it
+ reaches the end of the file, but will instead start monitoring it
+ and will start reading data again whenever a change is detected. The
+ new "waiting-data" and "done-waiting-data" signals keep the
+ application informed about the current state.
+
+- gtksink, gtkglsink:
+
+ - scroll event support: forwarded as navigation events into the
+ pipeline
+ - "video-aspect-ratio-override" property to force a specific
+ aspect ratio
+ - "rotate-method" property and support automatic rotation based on
+ image tags
+
+- identity: new "stats" property allows applications to retrieve the
+ number of bytes and buffers that have passed through so far.
+
+- interlace: add support for more formats, esp 10-bit, 12-bit and
+ 16-bit ones
+
+- jack: new "low-latency" property for automatic latency-optimized
+ setting and "port-names" property to select ports explicitly
+
+- jpegdec: support output conversion to RGB using libjpeg-turbo (for
+ certain input files)
+
+- line21dec:
+
+ - "mode" property to control whether and how detected closed
+ captions should be inserted in the list of existing close
+ caption metas on the input frame (if any): add, drop, or
+ replace.
+ - "ntsc-only" property to only look for captions if video has NTSC
+ resolution
+
+- line21enc: new "remove-caption-meta" to remove metas from output
+ buffers after encoding the captions into the video data; support for
+ CDP closed captions
+
+- matroskademux, matroskamux: Add support for ffv1, a lossless
+ intra-frame video coding format.
+
+- matroskamux: accept in-band SPS/PPS/VPS for H264 and H265
+ (i.e. stream-format avc3 and hev1) which allows on-the-fly
+ profile/level/resolution changes.
+
+- matroskamux: new "cluster-timestamp-offset" property, useful for use
+ cases where the container timestamps should map to some absolute
+ wall clock time, for example.
+
+- rtpsrc: add "caps" property to allow explicit setting of the caps
+ where needed
+
+- mpegts: support SCTE-35 passthrough via new "send-scte35-events"
+ property on MPEG-TS demuxer tsdemux. When enabled, SCTE 35 sections
+ (eg ad placement opportunities) are forwarded as events donwstream
+ where they can be picked up again by mpegtsmux. This required a
+ semantic change in the SCTE-35 section API: timestamps are now in
+ running time instead of muxer pts.
+
+- tsdemux: Handle PCR-less MPEG-TS streams; more robust timestamp
+ handling in certain corner cases and for poorly muxed streams.
+
+- mpegtsmux:
+
+ - More conformance improvements to make MPEG-TS analyzers happy:
+ - PCR timing accuracy: Improvements to the way mpegtsmux
+ outputs PCR observations in CBR mode, so that a PCR
+ observation is always inserted when needed, so that we never
+ miss the configured pcr-interval, as that triggers various
+ MPEG-TS analyser errors.
+ - Improved PCR/SI scheduling
+ - Don’t write PCR until PAT/PMT are output to make sure streams
+ start cleanly with a PAT/PMT.
+ - Allow overriding the automatic PMT PID selection via
+ application-supplied PMT_%d fields in the prog-map
+ structure/property.
+
+- mp4mux:
+
+ - new "first-moov-then-finalise" mode for fragmented output where
+ the output will start with a self-contained moov atom for the
+ first fragment, and then produce regular fragments. Then at the
+ end when the file is finalised, the initial moov is invalidated
+ and a new moov is written covering the entire file. This way the
+ file is a “fragmented mp4” file while it is still being written
+ out, and remains playable at all times, but at the end it is
+ turned into a regular mp4 file (with former fragment headers
+ remaining as unused junk data in the file).
+ - support H.264 avc3 and H.265 hvc1 stream formats as input where
+ the codec data is signalled in-band inside the bitstream instead
+ of caps/file headers.
+ - support profile/level/resolution changes for H264/H265 input
+ streams (i.e. codec data changing on the fly). Each codec_data
+ is put into its own SampleTableEntry inside the stsd, unless the
+ input is in avc3 stream format in which case it’s written
+ in-band an not in the headers.
+
+- multifilesink: new ""min-keyframe-distance"" property to make
+ minimum distance between keyframes in next-file=key-frame mode
+ configurable instead of hard-coding it to 10 seconds.
+
+- mxfdemux has seen a big refactoring to support non-frame wrappings
+ and more accurate timestamp/seek handling for some formats
+
+- msdk plugin for hardware-accelerated video encoding and decoding
+ using the Intel Media SDK:
+
+ - oneVPL support (Intel oneAPI Video Processing Library)
+ - AV1 decoding support
+ - H.264 decoder now supports constrained-high and progressive-high
+ profiles
+ - H.264 encoder:
+ - more configuration options (properties):
+ "intra-refresh-type", "min-qp" , "max-qp", "p-pyramid",
+ "dblk-idc"
+ - H.265 encoder:
+ - can output main-still-picture profile
+ - now inserts HDR SEIs (mastering display colour volume and
+ content light level)
+ - more configuration options (properties):
+ "intra-refresh-type", "min-qp" , "max-qp", "p-pyramid",
+ "b-pyramid", "dblk-idc", "transform-skip"
+ - support for RGB 10bit format
+ - External bitrate control in encoders
+ - Video post proc element msdkvpp gained support for 12-bit pixel
+ formats P012_LE, Y212_LE and Y412_LE
+
+- nvh264sldec: interlaced stream support
+
+- openh264enc: support main, high, constrained-high and
+ progressive-high profiles
+
+- openjpeg: support for multithreaded decoding and encoding
+
+- rtspsrc: now supports IPv6 also for tunneled mode (RTSP-over-HTTP);
+ new "ignore-x-server-reply" property to ignore the
+ x-server-ip-address server header reply in case of HTTP tunneling,
+ as it is often broken.
+
+- souphttpsrc: Runtime compatibility support for libsoup2 and
+ libsoup3. libsoup3 is the latest major version of libsoup, but
+ libsoup2 and libsoup3 can’t co-exist in the same process because
+ there is no namespacing or versioning for GObject types. As a
+ result, it would be awkward if the GStreamer souphttpsrc plugin
+ linked to a specific version of libsoup, because it would only work
+ with applications that use the same version of libsoup. To make this
+ work, the soup plugin now tries to determine the libsoup version
+ used by the application (and its other dependencies) at runtime on
+ systems where GStreamer is linked dynamically. libsoup3 support is
+ still considered somewhat experimental at this point.
+
+- srtsrc, srtsink: add signals for the application to accept/reject
+ incoming connections
+
+- timeoverlay: new elapsed-running-time time mode which shows the
+ running time since the first running time (and each flush-stop).
+
+- udpsrc: new timestamping mode to retrieve packet receive timestamps
+ from the kernel via socket control messages (SO_TIMESTAMPNS) on
+ supported platforms
+
+- uritranscodebin: new setup-source and element-setup signals for
+ applications to configure elements used
+
+- v4l2codecs plugin gained support for 4x4 and 32x32 tile formats
+ enabling some platforms or direct renders. Important memory usage
+ improvement.
+
+- v4l2slh264dec now implements the final Linux uAPI as shipped on
+ Linux 5.11 and later.
+
+- valve: add "drop-mode" property and provide two new modes of
+ operation: in drop-mode=forward-sticky-events sticky events
+ (stream-start, segment, tags, caps, etc.) are forwarded downstream
+ even when dropping is enabled; drop-mode=transform-to-gap will in
+ addition also convert buffers into gap events when dropping is
+ enabled, which lets downstream elements know that time is advancing
+ and might allow for preroll in many scenarios. By default all events
+ and all buffers are dropped when dropping is enabled, which can
+ cause problems with caps negotiation not progressing or branches not
+ prerolling when dropping is enabled.
+
+- videocrop: support for many more pixel formats, e.g. planar YUV
+ formats with > 8bits and GBR* video formats; can now also accept
+ video not backed by system memory as long as downstream supports the
+ GstCropMeta
+
+- videotestsrc: new smpte-rp-219 pattern for SMPTE75 RP-219 conformant
+ color bars
+
+- vp8enc: finish support for temporal scalability: two new properties
+ ("temporal-scalability-layer-flags",
+ "temporal-scalability-layer-sync-flags") and a unit change on the
+ "temporal-scalability-target-bitrate" property (now expects bps);
+ also make temporal scalability details available to RTP payloaders
+ as buffer metadata.
+
+- vp9enc: new properties to tweak encoder performance:
+
+ - "aq-mode" to configure adaptive quantization modes
+ - "frame-parallel-decoding" to configure whether to create a
+ bitstream that reduces decoding dependencies between frames
+ which allows staged parallel processing of more than one video
+ frames in the decoder. (Defaults to TRUE)
+ - "row-mt", "tile-columns" and "tile-rows" so multithreading can
+ be enabled on a per-tile basis, instead of on a per tile-column
+ basis. In combination with the new "tile-rows" property, this
+ allows the encoder to make much better use of the available CPU
+ power.
+
+- vp9dec, vp9enc: add support for 10-bit 4:2:0 and 4:2:2 YUV, as well
+ as 8-bit 4:4:4
+
+- vp8enc, vp9enc now default to “good quality” for the deadline
+ property rather then “best quality”. Having the deadline set to best
+ quality causes the encoder to be absurdly slow, most real-life users
+ will prefer good-enough quality with better performance instead.
+
+- wpesrc:
+
+ - implement audio support: a new sometimes source pad will be
+ created for each audio stream created by the web engine.
+ - move wpesrc to wpevideosrc and add a wrapper bin wpesrc to also
+ support audio
+ - also handles web:// URIs now (same as cefsrc)
+ - post messages with the estimated load progress on the bus
-Plugin and library moves
+- x265enc: add negative DTS support, which means timestamps are now
+ offset by 1h same as with x264enc
+
+RTP Payloaders and Depayloaders
+
+- rtpisacpay, rtpisacdepay: new RTP payloader and depayloader for iSAC
+ audio codec
+
+- rtph264depay:
+
+ - new "request-keyframe" property to make the depayloader
+ automatically request a new keyframe from the sender on packet
+ loss, consistent with the new property on rtpvp8depay.
+ - new "wait-for-keyframe" property to make depayloader wait for a
+ new keyframe at the beginning and after packet loss (only
+ effective if the depayloader outputs AUs), consistent with the
+ existing property on rtpvp8depay.
+
+- rtpopuspay, rtpopusdepay: support libwebrtc-compatible multichannel
+ audio in addition to the previously supported multichannel audio
+ modes
+
+- rtpopuspay: add DTX (Discontinuous Transmission) support
+
+- rtpvp8depay: new "request-keyframe" property to make the depayloader
+ automatically request a new keyframe from the sender on packet loss.
+
+- rtpvp8pay: temporal scaling support
-- this section will be filled in in due course
+- rtpvp9depay: Improved SVC handling (aggregate all layers)
+
+RTP Infrastructure
+
+- rtpst2022-1-fecdec, rtpst2022-1-fecenc: new elements providing SMPTE
+ 2022-1 2-D Forward Error Correction. More details in Mathieu’s blog
+ post.
+
+- rtpreddec: BUNDLE support
+
+- rtpredenc, rtpulpfecenc: add support for Transport-wide Congestion
+ Control (TWCC)
+
+- rtpsession: new "twcc-feedback-interval" property to allow RTCP TWCC
+ reports to be scheduled on a timer instead of per marker-bit.
+
+Plugin and library moves
- There were no plugin moves or library moves in this cycle.
The following elements or plugins have been removed:
-- this section will be filled in in due course
-
- The ofa audio fingerprinting plugin has been removed. The MusicIP
database has been defunct for years so this plugin is likely neither
useful nor used by anyone.
Miscellaneous API additions
-- this section will be filled in in due course
+Core
+
+- gst_buffer_new_memdup() is a convenience function for the
+ widely-used gst_buffer_new_wrapped(g_memdup(data,size),size)
+ pattern.
+
+- gst_caps_features_new_single() creates a new single GstCapsFeatures,
+ avoiding the need to use the vararg function with NULL terminator
+ for simple cases.
+
+- gst_element_type_set_skip_documentation() can be used by plugins to
+ signal that certain elements should not be included in the GStreamer
+ plugin documentation. This is useful for plugins where elements are
+ registered dynamically based on hardware capabilities and/or where
+ the available plugins and properties vary from system to system.
+ This is used in the d3d11 plugin for example to ensure that only the
+ list of default elements is advertised in the documentation.
+
+- gst_type_find_suggest_empty_simple() is a new convenience function
+ for typefinders for cases where there’s only a media type and no
+ other fields.
+
+- New API to create elements and set properties at construction time,
+ which is not only convenient, but also allows GStreamer elements to
+ have construct-only properties: gst_element_factory_make_full(),
+ gst_element_factory_make_valist(),
+ gst_element_factory_make_with_properties(),
+ gst_element_factory_create_full(),
+ gst_element_factory_create_valist(),
+ gst_element_factory_create_with_properties().
+
+- GstSharedTaskPool: new “shared” task pool subclass with slightly
+ different default behaviour than the existing GstTaskPool which
+ would create unlimited number of threads for new tasks. The shared
+ taskpool creates up to N threads (default: 1) and then distributes
+ pending tasks to those threads round-robin style, and blocks if no
+ thread is available. It is possible to join tasks. This can be used
+ by plugins to implement simple multi-threaded processing and is used
+ for the new multi-threaded video conversion and compositing done in
+ GstVideoAggregator, videoconverter and compositor.
+
+Plugins Base Utils library
+
+- GstDiscoverer:
+
+ - gst_discoverer_container_info_get_tags() was added to retrieve
+ global/container tags (vs. per-stream tags). Per-Stream tags can
+ be retrieved via the existing
+ gst_discoverer_stream_info_get_tags().
+ gst_discoverer_info_get_tags(), which for many files returns a
+ confusing mix of stream and container tags, has been deprecated
+ in favour of the container/stream-specific functions.
+ - gst_discoverer_stream_info_get_stream_number() returns a unique
+ integer identifier for a given stream within the given
+ GstDiscoverer context. (If this matches the stream number inside
+ the container bitstream that’s by coincidence and not by
+ design.)
+
+- gst_pb_utils_get_caps_description_flags() can be used to query
+ whether certain caps represent a container, audio, video, image,
+ subtitles, tags, or something else. This only works for formats
+ known to GStreamer.
+
+- gst_pb_utils_get_file_extension_from_caps() returns a possible file
+ extension for given caps.
+
+- gst_codec_utils_h264_get_profile_flags_level(): Parses profile,
+ flags, and level from H264 AvcC codec_data. The format of H264 AVCC
+ extradata/sequence_header is documented in the ITU-T H.264
+ specification section 7.3.2.1.1 as well as in ISO/IEC 14496-15
+ section 5.3.3.1.2.
+
+- gst_codec_utils_caps_get_mime_codec() to convert caps to a RFC 6381
+ compatible MIME codec string codec. Useful for providing the codecs
+ field inside the Content-Type HTTP header for containerized formats,
+ such as mp4 or matroska.
+
+GStreamer OpenGL integration library and plugins
+
+- glcolorconvert: added suppport for converting the video formats
+ A420, AV12, BGR, BGRA, RGBP and BGRP.
+
+- Added support to GstGLBuffer for persistent buffer mappings where a
+ Pixel Buffer Object (PBO) can be mapped by both the CPU and the GPU.
+ This removes a memcpy() when uploading textures or vertices
+ particularly when software decoders (e.g. libav) are direct
+ rendering into our memory. Improves transfer performance
+ significantly. Requires OpenGL 4.4, GL_ARB_buffer_storage or
+ GL_EXT_buffer_storage
+
+- Added various helper functions for handling 4x4 matrices of affine
+ transformations as used by GstVideoAffineTransformationMeta.
+
+- Add support to GstGLContext for allowing the application to control
+ the config (EGLConfig, GLXConfig, etc) used when creating the OpenGL
+ context. This allows the ability to choose between RGB16 or RGB10A2
+ or RGBA8 back/front buffer configurations that were previously
+ hardcoded. GstGLContext also supports retrieving the configuration
+ it was created with or from an externally provide OpenGL context
+ handle. This infrastructure is also used to create a compatible
+ config from an application/externally provided OpenGL context in
+ order to improve compatibility with other OpenGL frameworks and GUI
+ toolkits. A new environment variable GST_GL_CONFIG was also added to
+ be able to request a specific configuration from the command line.
+ Note: different platforms will have different functionality
+ available.
+
+- Add support for choosing between EGL and WGL at runtime when running
+ on Windows. Previously this was a build-time switch. Allows use in
+ e.g. Gtk applications on Windows that target EGL/ANGLE without
+ recompiling GStreamer. gst_gl_display_new_with_type() can be used by
+ applications to choose a specific display type to use.
+
+- Build fixes to explicitly check for Broadcom-specific libraries on
+ older versions of the Raspberry Pi platform. The Broadcom OpenGL ES
+ and EGL libraries have different filenames. Using the vc4 Mesa
+ driver on the Raspberry Pi is not affected.
+
+- Added support to glupload and gldownload for transferring RGBA
+ buffers using the memory:NVMM available on the Nvidia Tegra family
+ of embedded devices.
+
+- Added support for choosing libOpenGL and libGLX as used in a GLVND
+ environment on unix-based platforms. This allows using desktop
+ OpenGL and EGL without pulling in any GLX symbols as would be
+ required with libGL.
+
+Video library
+
+- New raw video formats:
+
+ - AV12 (NV12 with alpha plane)
+ - RGBP and BGRP (planar RGB formats)
+ - ARGB64 variants with specified endianness instead of host
+ endianness:
+ - ARGB64_LE, ARGB64_BE
+ - RGBA64_BE, RGBA64_LE
+ - BGRA64_BE, BGRA64_LE
+ - ABGR64_BE, ABGR64_LE
+
+- gst_video_orientation_from_tag() is new convenience API to parse the
+ image orientation from a GstTagList.
+
+- GstVideoDecoder subframe support (see below)
+
+- GstVideoCodecState now also carries some HDR metadata
+
+- Ancillary video data: implement transform functions for AFD/Bar
+ metas, so they will be forwarded in more cases
+
+MPEG-TS library
+
+This library only handles section parsing and such, see above for
+changes to the actual mpegtsmux and mpegtsdemux elements.
+
+- many additions and improvements to SCTE-35 section parsing
+- new API for fetching extended descriptors:
+ gst_mpegts_find_descriptor_with_extension()
+- add support for SIT sections (Selection Information Tables)
+- expose event-from-section constructor gst_event_new_mpegts_section()
+- parse Audio Preselection Descriptor needed for Dolby AC-4
+
+GstWebRTC library + webrtcbin
+
+- Change the way in which sink pads and transceivers are matched
+ together to support easier usage. If a pad is created without a
+ specific index (i.e. using sink_%u as the pad template), then an
+ available compatible transceiver will be searched for. If a specific
+ index is requested (i.e. sink_1) then if a transceiver for that
+ m-line already exists, that transceiver must match the new sink pad
+ request. If there is no transceiver available in either scenario, a
+ new transceiver is created. If a mixture of both sink_1 and sink_%u
+ requests result in an impossible situation, an error will be
+ produced at pad request time or from create offer/answer.
+
+- webrtcbin now uses regular ICE nomination instead of libnice’s
+ default of aggressive ICE nomination. Regular ICE nomination is the
+ default recommended by various relevant standards and improves
+ connectivity in specific network scenarios.
+
+- Add support for limiting the port range used for RTP with the
+ addition of the min-rtp-port and max-rtp-port properties on the ICE
+ object.
+
+- Expose the SCTP transport as a property on webrtcbin to more closely
+ match the WebRTC specification.
+
+- Added support for taking into account the data channel transport
+ state when determining the value of the "connection-state" property.
+ Previous versions of the WebRTC spec did not include the data
+ channel state when computing this value.
+
+- Add configuration for choosing the size of the underlying sockets
+ used for transporting media data
+
+- Always advertise support for the transport-cc RTCP feedback protocol
+ as rtpbin supports it. For full support, the configured caps (input
+ or through codec-preferences) need to include the relevant RTP
+ header extension.
+
+- Numerous fixes to caps and media handling to fail-fast when an
+ incompatible situation is detected.
+
+- Improved support for attaching the required media after a remote
+ offer has been set.
+
+- Add support for dynamically changing the amount of FEC used for a
+ particular stream.
+
+- webrtcbin now stops further SDP processing at the first error it
+ encounters.
+
+- Completed support for either local or the remote closing a data
+ channel.
+
+- Various fixes when performing BUNDLEing of the media streams in
+ relation to RTX and FEC usage.
+
+- Add support for writing out QoS DSCP marking on outgoing packets to
+ improve reliability in some network scenarios.
+
+- Improvements to the statistics returned by the get-stats signal
+ including the addition of the raw statistics from the internal
+ RTPSource, the TWCC stats when available.
+
+- The webrtc library does not expose any objects anymore with public
+ fields. Instead properties have been added to replace that
+ functionality. If you are accessing such fields in your application,
+ switch to the corresponding properties.
+
+GstCodecs and Video Parsers
+
+- Support for render delays to improve throughput across all CODECs
+ (used with NVDEC and V4L2).
+- lots of improvements to parsers and the codec parsing decoder base
+ classes (H264, H265, VP8, VP9, AV1, MPEG-2) used for various
+ hardware-accelerate decoder APIs.
+
+Bindings support
+
+- gst_allocation_params_new() allocates a GstAllocationParams struct
+ on the heap. This should only be used by bindings (and freed via
+ gst_allocation_params_free() then). In C code you would allocate
+ this on the stack and only init it in place.
+
+- gst_debug_log_literal() can be used to log a string to the debug log
+ without going through any printf format expansion and associated
+ overhead. This is mostly useful for bindings such as the Rust
+ bindings which may have done their own formatting already .
+
+- Provide non-inlined versions of refcounting APIs for various
+ GStreamer mini objects, so that they can be consumed by bindings
+ (e.g. gstreamer-sharp): gst_buffer_ref, gst_buffer_unref,
+ gst_clear_buffer, gst_buffer_copy, gst_buffer_replace,
+ gst_buffer_list_ref, gst_buffer_list_unref, gst_clear_buffer_list,
+ gst_buffer_list_copy, gst_buffer_list_replace, gst_buffer_list_take,
+ gst_caps_ref, gst_caps_unref, gst_clear_caps, gst_caps_replace,
+ gst_caps_take, gst_context_ref, gst_context_unref, gst_context_copy,
+ gst_context_replace, gst_event_replace, gst_event_steal,
+ gst_event_take, gst_event_ref, gst_event_unref, gst_clear_event,
+ gst_event_copy, gst_memory_ref, gst_memory_unref, gst_message_ref,
+ gst_message_unref, gst_clear_message, gst_message_copy,
+ gst_message_replace, gst_message_take, gst_promise_ref,
+ gst_promise_unref, gst_query_ref, gst_query_unref, gst_clear_query,
+ gst_query_copy, gst_query_replace, gst_query_take, gst_sample_ref,
+ gst_sample_unref, gst_sample_copy, gst_tag_list_ref,
+ gst_tag_list_unref, gst_clear_tag_list, gst_tag_list_replace,
+ gst_tag_list_take, gst_uri_copy, gst_uri_ref, gst_uri_unref,
+ gst_clear_uri.
+
+- expose a GType for GstMiniObject
+
+- gst_device_provider_probe() now returns non-floating device object
+
+API Deprecations
+
+- gst_element_get_request_pad() has been deprecated in favour of the
+ newly-added gst_element_request_pad_simple() which does the exact
+ same thing but has a less confusing name that hopefully makes clear
+ that the function request a new pad rather than just retrieves an
+ already-existing request pad.
+
+- gst_discoverer_info_get_tags(), which for many files returns a
+ confusing mix of stream and container tags, has been deprecated in
+ favour of the container-specific and stream-specific functions,
+ gst_discoverer_container_info_get_tags() and
+ gst_discoverer_stream_info_get_tags().
+
+- gst_video_sink_center_rect() was deprecated in favour of the more
+ generic newly-added gst_video_center_rect().
+
+- The GST_MEMORY_FLAG_NO_SHARE flag has been deprecated, as it tends
+ to cause problems and prevents sub-buffering. If pooling or lifetime
+ tracking is required, memories should be allocated through a custom
+ GstAllocator instead of relying on the lifetime of the buffers the
+ memories were originally attached to, which is fragile anyway.
+
+- The GstPlayer high-level playback library is being replaced with the
+ new GstPlay library (see above). GstPlayer should be considered
+ deprecated at this point and will be marked as such in the next
+ development cycle. Applications should be ported to GstPlay.
+
+- Gstreamer Editing Services: ges_video_transition_set_border(),
+ ges_video_transition_get_border()
+ ges_video_transition_set_inverted()
+ ges_video_transition_is_inverted() have been deprecated, use
+ ges_timeline_element_set_children_properties() instead.
Miscellaneous performance, latency and memory optimisations
-- this section will be filled in in due course
+More video conversion fast paths
-Miscellaneous other changes and enhancements
+- v210 ↔ I420, YV12, Y42B, UYVY and YUY2
+- A420 → RGB
+
+Less jitter when waiting on the system clock
+
+- Better system clock wait accuracy, less jitter: where available,
+ clock_nanosleep is used for higher accuracy for waits below 500
+ usecs, and waits below 2ms will first use the regular waiting system
+ and then clock_nanosleep for the remainder. The various wait
+ implementation have a latency ranging from 50 to 500+ microseconds.
+ While this is not a major issue when dealing with a low number of
+ waits per second (for ex: video), it does introduce a non-negligible
+ jitter for synchronisation of higher packet rate systems.
+
+Video decoder subframe support
-- this section will be filled in in due course
+- The GstVideoDecoder base class gained API to process input at the
+ sub-frame level. That way video decoders can start decoding slices
+ before they have received the full input frame in its entirety (to
+ the extent this is supported by the codec, of course). This helps
+ with CPU utilisation and reduces latency.
-Tracing framework and debugging improvements
+- This functionality is now being used in the OpenJPEG JPEG 2000
+ decoder, the FFmpeg H.264 decoder (in case of NAL-aligned input) and
+ the OpenMAX H.264/H.265 decoders (in case of NAL-aligned input).
-- this section will be filled in in due course
+Miscellaneous other changes and enhancements
+
+- GstDeviceMonitor no longer fails to start just because one of the
+ device providers failed to start. That could happen for example on
+ systems where the pulseaudio device provider is installed, but
+ pulseaudio isn’t actually running but ALSA is used for audio
+ instead. In the same vein the device monitor now keeps track of
+ which providers have been started (via the new
+ gst_device_provider_is_started()) and only stops actually running
+ device providers when stopping the device monitor.
+
+- On embedded systems it can be useful to create a registry that can
+ be shared and read by multiple processes running as different users.
+ It is now possible to set the new GST_REGISTRY_MODE environment
+ variable to specify the file mode for the registry file, which by
+ default is set to be only user readable/writable.
+
+- GstNetClientClock will signal lost sync in case the remote time
+ resets (e.g. because device power cycles), by emitting the “synced”
+ signal with synced=FALSE parameter, so applications can take action.
+
+- gst_value_deserialize_with_pspec() allows deserialization with a
+ hint for what the target GType should be. This allows for example
+ passing arrays of flags through the command line or
+ gst_util_set_object_arg(), eg: foo="<bar,bar+baz>".
+
+- It’s now allowed to create an empty GstVideoOverlayComposition
+ without any rectangles by passing a NULL rectangle to
+ gst_video_overlay_composition_new(). This is useful for bindings and
+ simplifies application code in some places.
+
+Tracing framework, debugging and testing improvements
+
+- New factories tracer to list loaded elements (and other plugin
+ features). This can be useful to collect a list of elements needed
+ for an application, which then in turn can be used to create a
+ tailored minimal GStreamer build that contains just the elements
+ needed and nothing else.
+- New plugin-feature-loaded tracing hook for use by tracers like the
+ new factories tracer
+
+- GstHarness: Add gst_harness_set_live() so that harnesses can be set
+ to non-live and return is-live=false in latency queries if needed.
+ Default behaviour is to always return is-live=true in latency
+ queries.
+
+- navseek: new "hold-eos" property. When enabled, the element will
+ hold back an EOS event until the next keystroke (via navigation
+ events). This can be used to keep a video sink showing the last
+ frame of a video pipeline until a key is pressed instead of tearing
+ it down immediately on EOS.
+
+- New fakeaudiosink element: mimics an audio sink and can be used for
+ testing and CI pipelines on systems where no audio system is
+ installed or running. It differs from fakesink in that it only
+ support audio caps and syncs to the clock by default like a normal
+ audio sink. It also implements the GstStreamVolume interface like
+ most audio sinks do.
+
+- New videocodectestsink element for video codec conformance testing:
+ Calculates MD5 checksums for video frames and skips any padding
+ whilst doing so. Can optionally also write back the video data with
+ padding removed into a file for easy byte-by-byte comparison with
+ reference data.
Tools
-- this section will be filled in in due course
+gst-inspect-1.0
+
+- Can sort the list of plugins by passing --sort=name as command line
+ option
+
+gst-launch-1.0
+
+- will now error out on top-level properties that don’t exist and
+ which were silently ignored before
+- On Windows the high-resolution clock is enabled now, which provides
+ better clock and timer performance on Windows (see Windows section
+ below for more details).
+
+gst-play-1.0
+
+- New --start-position command line argument to start playback from
+ the specified position
+- Audio can be muted/unmuted in interactive mode by pressing the m
+ key.
+- On Windows the high-resolution clock is enabled now (see Windows
+ section below for more details)
+
+gst-device-monitor-1.0
+
+- New --include-hidden command line argument to also show “hidden”
+ device providers
+
+ges-launch-1.0
+
+- New interactive mode that allows seeking and such. Can be disabled
+ by passing the --no-interactive argument on the command line.
+- Option to forward tags
+- Allow using an existing clip to determine the rendering format (both
+ topology and profile) via new --profile-from command line argument.
GStreamer RTSP server
-- this section will be filled in in due course
+- GstRTSPMediaFactory gained API to disable RTCP
+ (gst_rtsp_media_factory_set_enable_rtcp(), "enable-rtcp" property).
+ Previously RTCP was always allowed for all RTSP medias. With this
+ change it is possible to disable RTCP completely, no matter if the
+ client wants to do RTCP or not.
+
+- Make a mount point of / work correctly. While not allowed by the
+ RTSP 2 spec, the RTSP 1 spec is silent on this and it is used in the
+ wild. It is now possible to use / as a mount path in
+ gst-rtsp-server, e.g. rtsp://example.com/ would work with this now.
+ Note that query/fragment parts of the URI are not necessarily
+ correctly handled, and behaviour will differ between various
+ client/server implementations; so use it if you must but don’t bug
+ us if it doesn’t work with third party clients as you’d hoped.
+
+- multithreading fixes (races, refcounting issues, deadlocks)
+
+- ONVIF audio backchannel fixes
+
+- ONVIF trick mode optimisations
+
+- rtspclientsink: new "update-sdp" signal that allows updating the SDP
+ before sending it to the server via ANNOUNCE. This can be used to
+ add additional metadata to the SDP, for example. The order and
+ number of medias must not be changed, however.
GStreamer VAAPI
-- this section will be filled in in due course
+- new AV1 decoder element (vaapiav1dec)
+
+- H264 decoder: handle stereoscopic 3D video with frame packing
+ arrangement SEI messages
+
+- H265 encoder: added Screen Content Coding extensions support
+
+- H265 decoder: gained MAIN_444_12 profile support (decoded to
+ Y412_LE), and 4:2:2 12-bits support (decoded to Y212_LE)
+
+- vaapipostproc: gained BT2020 color standard support
+
+- vaapidecode: now generates caps templates dynamically at runtime in
+ order to advertise actually supported caps instead of all
+ theoretically supported caps.
+
+- GST_VAAPI_DRM_DEVICE environment variable to force a specified DRM
+ device when a DRM display is used. It is ignored when other types of
+ displays are used. By default /dev/dri/renderD128 is used for DRM
+ display.
GStreamer OMX
-- this section will be filled in in due course
+- subframe support in H.264/H.265 decoders
GStreamer Editing Services and NLE
-- this section will be filled in in due course
+- framepositioner: new "operator" property to access blending modes in
+ the compositor
+- timeline: Implement snapping to markers
+- smart-mixer: Add support for d3d11compositor and glvideomixer
+- titleclip: add "draw-shadow" child property
+- ges:// URI support to define a timeline from a description.
+- command-line-formatter
+ - Add track management to timeline description
+ - Add keyframe support
+- ges-launch-1.0:
+ - Add an interactive mode where we can seek etc…
+ - Add option to forward tags
+ - Allow using an existing clip to determine the rendering format
+ (both topology and profile) via new --profile-from command line
+ argument.
+- Fix static build
GStreamer validate
-- this section will be filled in in due course
+- report: Add a way to force backtraces on reports even if not a
+ critical issue (GST_VALIDATE_ISSUE_FLAGS_FORCE_BACKTRACE)
+- Add a flag to gst_validate_replace_variables_in_string() allow
+ defining how to resolve variables in structs
+- Add gst_validate_bin_monitor_get_scenario() to get the bin monitor
+ scenario, which is useful for applications that use Validate
+ directly.
+- Add an expected-values parameter to wait, message-type=XX allowing
+ more precise filtering of the message we are waiting for.
+- Add config file support: each test can now use a config file for the
+ given media file used to test.
+- Add support to check properties of object properties
+- scenario: Add an "action-done" signal to signal when an action is
+ done
+- scenario: Add a "run-command" action type
+- scenario: Allow forcing running action on idle from scenario file
+- scenario: Allow iterating over arrays in foreach
+- scenario: Rename ‘interlaced’ action to ‘non-blocking’
+- scenario: Add a non-blocking flag to the wait signal
GStreamer Python Bindings
-- this section will be filled in in due course
+- Fixes for Python 3.10
+- Various build fixes
+- at least one known breaking change caused by g-i annotation changes
+ (see below)
GStreamer C# Bindings
-- this section will be filled in in due course
+- Fix GstDebugGraphDetails enum
+- Updated to latests GtkSharp
+- Updated to include GStreamer 1.20 API
GStreamer Rust Bindings and Rust Plugins
-The GStreamer Rust bindings are released separately with a different
-release cadence that’s tied to gtk-rs, but the latest release has
-already been updated for the upcoming new GStreamer 1.20 API.
-
-gst-plugins-rs, the module containing GStreamer plugins written in Rust,
-has also seen lots of activity with many new elements and plugins.
-
-What follows is a list of elements and plugins available in
-gst-plugins-rs, so people don’t miss out on all those potentially useful
-elements that have no C equivalent.
-
-- FIXME: add new elements
-
-Rust audio plugins
-
-- audiornnoise: New element for audio denoising which implements the
- noise removal algorithm of the Xiph RNNoise library, in Rust
-- rsaudioecho: Port of the audioecho element from gst-plugins-good
- rsaudioloudnorm: Live audio loudness normalization element based on
- the FFmpeg af_loudnorm filter
-- claxondec: FLAC lossless audio codec decoder element based on the
- pure-Rust claxon implementation
-- csoundfilter: Audio filter that can use any filter defined via the
- Csound audio programming language
-- lewtondec: Vorbis audio decoder element based on the pure-Rust
- lewton implementation
-
-Rust video plugins
-
-- cdgdec/cdgparse: Decoder and parser for the CD+G video codec based
- on a pure-Rust CD+G implementation, used for example by karaoke CDs
-- cea608overlay: CEA-608 Closed Captions overlay element
-- cea608tott: CEA-608 Closed Captions to timed-text (e.g. VTT or SRT
- subtitles) converter
-- tttocea608: CEA-608 Closed Captions from timed-text converter
-- mccenc/mccparse: MacCaption Closed Caption format encoder and parser
-- sccenc/sccparse: Scenarist Closed Caption format encoder and parser
-- dav1dec: AV1 video decoder based on the dav1d decoder implementation
- by the VLC project
-- rav1enc: AV1 video encoder based on the fast and pure-Rust rav1e
- encoder implementation
-- rsflvdemux: Alternative to the flvdemux FLV demuxer element from
- gst-plugins-good, not feature-equivalent yet
-- rsgifenc/rspngenc: GIF/PNG encoder elements based on the pure-Rust
- implementations by the image-rs project
-
-Rust text plugins
-
-- textwrap: Element for line-wrapping timed text (e.g. subtitles) for
- better screen-fitting, including hyphenation support for some
- languages
-
-Rust network plugins
-
-- reqwesthttpsrc: HTTP(S) source element based on the Rust
- reqwest/hyper HTTP implementations and almost feature-equivalent
- with the main GStreamer HTTP source souphttpsrc
-- s3src/s3sink: Source/sink element for the Amazon S3 cloud storage
-- awstranscriber: Live audio to timed text transcription element using
- the Amazon AWS Transcribe API
-
-Generic Rust plugins
-
-- sodiumencrypter/sodiumdecrypter: Encryption/decryption element based
- on libsodium/NaCl
-- togglerecord: Recording element that allows to pause/resume
- recordings easily and considers keyframe boundaries
-- fallbackswitch/fallbacksrc: Elements for handling potentially
- failing (network) sources, restarting them on errors/timeout and
- showing a fallback stream instead
-- threadshare: Set of elements that provide alternatives for various
- existing GStreamer elements but allow to share the streaming threads
- between each other to reduce the number of threads
-- rsfilesrc/rsfilesink: File source/sink elements as replacements for
- the existing filesrc/filesink elements
+- The GStreamer Rust bindings are released separately with a different
+ release cadence that’s tied to gtk-rs, but the latest release has
+ already been updated for the upcoming new GStreamer 1.20 API (v1_20
+ feature).
+
+- gst-plugins-rs, the module containing GStreamer plugins written in
+ Rust, has also seen lots of activity with many new elements and
+ plugins. See the New Elements section above for a list of new Rust
+ elements.
Build and Dependencies
- Meson 0.59 or newer is required to build GStreamer now.
-- FIXME: this section will be filled in in due course
+- The GLib requirement has been bumped to GLib 2.56 or newer (from
+ March 2018).
+
+- The wpe plugin now requires wpe >= 2.28 and wpebackend-fdo >= 1.8
Explicit opt-in required for build of certain plugins with (A)GPL dependencies
See Building plugins with (A)GPL-licensed dependencies for more details
and a non-exhaustive list of plugins affected.
-gst-build: replaced by Monorepo
+gst-build: replaced by mono repository
-- this section will be filled in in due course
-
-- FIXME: describe + link to Monorepo FAQ
+See mono repository section above and the GStreamer mono repository FAQ.
Cerbero
on platforms where dependencies are not readily available, such as
Windows, Android, iOS and macOS.
-General improvements
+General Cerbero improvements
-- this section will be filled in in due course
+- Plugin removed: libvisual
+- New plugins: rtpmanagerbad and rist
-macOS / iOS
+macOS / iOS specific Cerbero improvements
-- this section will be filled in in due course
+- XCode 12 support
+- macOS OS release support is now future-proof, similar to iOS
+- macOS Apple Silicon (ARM64) cross-compile support has been added
+- macOS Apple Silicon (ARM64) native support is currently experimental
-Windows
+Windows specific Cerbero improvements
-- this section will be filled in in due course
+- Visual Studio 2022 support has been added
+- bootstrap is faster since it requires building fewer build-tools
+ recipes on Windows
+- package is faster due to better scheduling of recipe stages and
+ elimination of unnecessary autotools regeneration
+- The following plugins are no longer built on Windows:
+ - a52dec (another decoder is still available in libav)
+ - dvdread
+ - resindvd
Windows MSI installer
-- this section will be filled in in due course
+- no major changes
-Linux
+Linux specific Cerbero improvements
-- this section will be filled in in due course
+- Fedora, Debian OS release support is now more future-proof
+- Amazon Linux 2 support has been added
-Android
+Android specific Cerbero improvements
-- this section will be filled in in due course
+- no major changes
Platform-specific changes and improvements
Android
-- this section will be filled in in due course
+- No major changes
macOS and iOS
-- this section will be filled in in due course
+- applemedia: add ProRes support to vtenc and vtdec
Windows
-- this section will be filled in in due course
+- On Windows the high-resolution clock is enabled now in the
+ gst-launch-1.0 and gst-play-1.0 command line tools, which provides
+ better clock and timer performance on Windows, at the cost of higher
+ power consumption. By default, without the high-resolution clock
+ enabled, the timer precision on Windows is system-dependent and may
+ be as bad as 15ms which is not good enough for many multimedia
+ applications. Developers may want to do the same in their Windows
+ applications if they think it’s a good idea for their application
+ use case, and depending on the Windows version they target. This is
+ not done automatically by GStreamer because on older Windows
+ versions (pre-Windows 10) this affects a global Windows setting and
+ also there’s a power consumption vs. performance trade-off that may
+ differ from application to application.
+
+- dxgiscreencapsrc now supports resolution changes
+
+- The wasapi2 audio plugin was rewritten and now has a higher rank
+ than the old wasapi plugin since it has a number of additional
+ features such as automatic stream routing, and no
+ known-but-hard-to-fix issues. The plugin is always built if the
+ Windows 10 SDK is available now.
+
+- The wasapi device providers now detect and notify dynamic device
+ additions/removals
+
+- d3d11screencapturesrc: new desktop capture element, including
+ GstDeviceProvider implementation to enumerate/select target monitors
+ for capture.
+
+- Direct3D11/DXVA decoder now supports AV1 and MPEG2 codecs
+ (d3d11av1dec, d3d11mpeg2dec)
+
+- VP9 decoding got more reliable and stable thanks to a newly written
+ codec parser
+
+- Support for decoding interlaced H.264/AVC streams
+
+- Hardware-accelerated video deinterlacing (d3d11deinterlace) and
+ video mixing (d3d11compositor)
+
+- Video mixing with the Direct3D11 API (d3d11compositor)
+
+- MediaFoundation API based hardware encoders gained the ability to
+ receive Direct3D11 textures as an input
+
+- Seungha’s blog post “GStreamer ❤ Windows: A primer on the cool stuff
+ you’ll find in the 1.20 release” describes many of the
+ Windows-related improvements in more detail
Linux
-- this section will be filled in in due course
+- bluez: LDAC Bluetooth audio codec support in a2dpsink and avdtpsink,
+ as well as an LDAC RTP payloader (rtpldacpay) and an LDAC audio
+ encoder (ldacenc)
+
+- kmssink: gained support for NV24, NV61, RGB16/BGR16 formats;
+ auto-detect NVIDIA Tegra driver
Documentation improvements
-- this section will be filled in in due course
+- hardware-accelerated GPU plugins will now no longer always list all
+ the element variants for all available GPUs, since those are
+ system-dependent and it’s confusing for users to see those in the
+ documentation just because the GStreamer developer who generated the
+ docs had multiple GPUs to play with at the time. Instead just show
+ the default elements.
-Possibly Breaking Changes
+Possibly Breaking and Other Noteworthy Behavioural Changes
+
+- gst_parse_launch(), gst_parse_bin_from_description() and friends
+ will now error out when setting properties that don’t exist on
+ top-level bins. They were silently ignored before.
+
+- The GstWebRTC library does not expose any objects anymore with
+ public fields. Instead properties have been added to replace that
+ functionality. If you are accessing such fields in your application,
+ switch to the corresponding properties.
-- FIXME: this section will be filled in in due course
-- MPEG-TS SCTE-35 API changes (FIXME: flesh out)
-- gst_parse_launch() and friends now error out on non-existing
- properties on top-level bins where they would silently fail and
- ignore those before.
- playbin and uridecodebin now emit the source-setup signal before the
element is added to the bin and linked so that the source element is
already configured before any scheduling query comes in, which is
- useful for elements such as appsrc or giostreamsrc. (Merge Request)
+ useful for elements such as appsrc or giostreamsrc.
+
+- The source element inside urisourcebin (used inside uridecodebin3
+ which is used inside playbin3) is no longer called "source". This
+ shouldn’t affect anyone hopefully, because there’s a "setup-source"
+ signal to configure the source element and no one should rely on
+ names of internal elements anyway.
+
+- The vp8enc element now expects bps (bits per second) for the
+ "temporal-scalability-target-bitrate" property, which is consistent
+ with the "target-bitrate" property. Since additional configuration
+ is required with modern libvpx to make temporal scaling work anyway,
+ chances are that very few people will have been using this property
+
+- vp8enc and vp9enc now default to “good quality” for the "deadline"
+ property rather then “best quality”. Having the deadline set to best
+ quality causes the encoder to be absurdly slow, most real-life users
+ will want the good quality tradeoff instead.
+
+- The experimental GstTranscoder library API in gst-plugins-bad was
+ changed from a GObject signal-based notification mechanism to a
+ GstBus/message-based mechanism akin to GstPlayer/GstPlay.
+
+- MPEG-TS SCTE-35 API: semantic change for SCTE-35 splice commands:
+ timestamps passed by the application should be in running time now,
+ since users of the API can’t really be expected to predict the local
+ PTS of the muxer.
+
+- The GstContext used by souphttpsrc to share the session between
+ multiple element instances has changed. Previously it provided
+ direct access to the internal SoupSession object, now it only
+ provides access to an opaque, internal type. This change is
+ necessary because SoupSession is not thread-safe at all and can’t be
+ shared safely between arbitrary external code and souphttpsrc.
+
+- Python bindings: GObject-introspection related Annotation fixes have
+ led to a case of a GstVideo.VideoInfo-related function signature
+ changing in the Python bindings (possibly one or two other cases
+ too). This is for a function that should never have been exposed in
+ the first place though, so the bindings are being updated to throw
+ an exception in that case, and the correct replacement API has been
+ added in form of an override.
Known Issues
-- this section will be filled in in due course
-
-- There are a couple of known WebRTC-related regressions/blockers:
-
- - webrtc: DTLS setup with Chrome is broken
- - webrtcbin: First keyframe is usually lost
+- nothing in particular at this point (but also see possibly breaking
+ changes section above)
Contributors
-- this section will be filled in in due course
+Aaron Boxer, Adam Leppky, Adam Williamson, Alba Mendez, Alejandro
+González, Aleksandr Slobodeniuk, Alexander Vandenbulcke, Alex Ashley,
+Alicia Boya García, Andika Triwidada, Andoni Morales Alastruey, Andrew
+Wesie, Andrey Moiseev, Antonio Ospite, Antonio Rojas, Arthur Crippa
+Búrigo, Arun Raghavan, Ashley Brighthope, Axel Kellermann, Baek, Bastien
+Nocera, Bastien Reboulet, Benjamin Gaignard, Bing Song, Binh Truong,
+Biswapriyo Nath, Brad Hards, Brad Smith, Brady J. Garvin, Branko
+Subasic, Camilo Celis Guzman, Chris Bass, ChrisDuncanAnyvision, Chris
+White, Corentin Damman, Daniel Almeida, Daniel Knobe, Daniel Stone,
+david, David Fernandez, David Keijser, David Phung, Devarsh Thakkar,
+Dinesh Manajipet, Dmitry Samoylov, Dmitry Shusharin, Dominique Martinet,
+Doug Nazar, Ederson de Souza, Edward Hervey, Emmanuel Gil Peyrot,
+Enrique Ocaña González, Ezequiel Garcia, Fabian Orccon, Fabrice
+Fontaine, Fernando Jimenez Moreno, Florian Karydes, Francisco Javier
+Velázquez-García, François Laignel, Frederich Munch, Fredrik Pålsson,
+George Kiagiadakis, Georg Lippitsch, Göran Jönsson, Guido Günther,
+Guillaume Desmottes, Guiqin Zou, Haakon Sporsheim, Haelwenn (lanodan)
+Monnier, Haihao Xiang, Haihua Hu, Havard Graff, He Junyan, Helmut
+Januschka, Henry Wilkes, Hosang Lee, Hou Qi, Ignacio Casal Quinteiro,
+Igor Kovalenko, Ilya Kreymer, Imanol Fernandez, Jacek Tomaszewski, Jade
+Macho, Jakub Adam, Jakub Janků, Jan Alexander Steffens (heftig), Jan
+Schmidt, Jason Carrete, Jason Pereira, Jay Douglass, Jeongki Kim, Jérôme
+Laheurte, Jimmi Holst Christensen, Johan Sternerup, John Hassell, John
+Lindgren, John-Mark Bell, Jonathan Matthew, Jordan Petridis, Jose
+Quaresma, Julian Bouzas, Julien, Kai Uwe Broulik, Kasper Steensig
+Jensen, Kellermann Axel, Kevin Song, Khem Raj, Knut Inge Hvidsten, Knut
+Saastad, Kristofer Björkström, Lars Lundqvist, Lawrence Troup, Lim Siew
+Hoon, Lucas Stach, Ludvig Rappe, Luis Paulo Fernandes de Barros, Luke
+Yelavich, Mads Buvik Sandvei, Marc Leeman, Marco Felsch, Marek Vasut,
+Marian Cichy, Marijn Suijten, Marius Vlad, Markus Ebner, Mart Raudsepp,
+Matej Knopp, Mathieu Duponchelle, Matthew Waters, Matthieu De Beule,
+Mengkejiergeli Ba, Michael de Gans, Michael Olbrich, Michael Tretter,
+Michal Dzik, Miguel Paris, Mikhail Fludkov, mkba, Nazar Mokrynskyi,
+Nicholas Jackson, Nicola Murino, Nicolas Dufresne, Niklas Hambüchen,
+Nikolay Sivov, Nirbheek Chauhan, Olivier Blin, Olivier Crete, Olivier
+Crête, Paul Goulpié, Per Förlin, Peter Boba, P H, Philippe Normand,
+Philipp Zabel, Pieter Willem Jordaan, Piotrek Brzeziński, Rafał
+Dzięgiel, Rafostar, raghavendra, Raghavendra, Raju Babannavar, Raleigh
+Littles III, Randy Li, Randy Li (ayaka), Ratchanan Srirattanamet, Raul
+Tambre, reed.lawrence, Ricky Tang, Robert Rosengren, Robert Swain, Robin
+Burchell, Roman Sivriver, R S Nikhil Krishna, Ruben Gonzalez, Ruslan
+Khamidullin, Sanchayan Maity, Scott Moreau, Sebastian Dröge, Sergei
+Kovalev, Seungha Yang, Sid Sethupathi, sohwan.park, Sonny Piers, Staz M,
+Stefan Brüns, Stéphane Cerveau, Stephan Hesse, Stian Selnes, Stirling
+Westrup, Théo MAILLART, Thibault Saunier, Tim, Timo Wischer, Tim-Philipp
+Müller, Tim Schneider, Tobias Ronge, Tom Schoonjans, Tulio Beloqui,
+tyler-aicradle, U. Artie Eoff, Ung, Val Doroshchuk, VaL Doroshchuk,
+Víctor Manuel Jáquez Leal, Vivek R, Vivia Nikolaidou, Vivienne
+Watermeier, Vladimir Menshakov, Will Miller, Wim Taymans, Xabier
+Rodriguez Calvar, Xavier Claessens, Xℹ Ruoyao, Yacine Bandou, Yinhang
+Liu, youngh.lee, youngsoo.lee, yychao, Zebediah Figura, Zhang yuankun,
+Zhang Yuankun, Zhao, Zhao Zhili, , Aleksandar Topic, Antonio Ospite,
+Bastien Nocera, Benjamin Gaignard, Brad Hards, Carlos Falgueras García,
+Célestin Marot, Corentin Damman, Corentin Noël, Daniel Almeida, Daniel
+Knobe, Danny Smith, Dave Piché, Dmitry Osipenko, Fabrice Fontaine,
+fjmax, Florian Zwoch, Guillaume Desmottes, Haihua Hu, Heinrich Kruger,
+He Junyan, Jakub Adam, James Cowgill, Jan Alexander Steffens (heftig),
+Jean Felder, Jeongki Kim, Jiri Uncovsky, Joe Todd, Jordan Petridis,
+Krystian Wojtas, Marc-André Lureau, Marcin Kolny, Marc Leeman, Mark
+Nauwelaerts, Martin Reboredo, Mathieu Duponchelle, Matthew Waters,
+Mengkejiergeli Ba, Michael Gruner, Nicolas Dufresne, Nirbheek Chauhan,
+Olivier Crête, Philippe Normand, Rafał Dzięgiel, Ralf Sippl, Robert
+Mader, Sanchayan Maity, Sangchul Lee, Sebastian Dröge, Seungha Yang,
+Stéphane Cerveau, Teh Yule Kim, Thibault Saunier, Thomas Klausner, Timo
+Wischer, Tim-Philipp Müller, Tobias Reineke, Tomasz Andrzejak, Trung Do,
+Tyler Compton, Ung, Víctor Manuel Jáquez Leal, Vivia Nikolaidou, Wim
+Taymans, wngecn, Wonchul Lee, wuchang li, Xavier Claessens, Xi Ruoyao,
+Yoshiharu Hirose, Zhao,
… and many others who have contributed bug reports, translations, sent
suggestions or helped testing.
1.20.0
-1.20.0 is scheduled to be released around October/November 2021.
+1.20.0 is scheduled to be released around early February 2022.
Schedule for 1.22
unstable development version leading up to the stable 1.22 release. The
development of 1.21/1.22 will happen in the git main branch.
-The plan for the 1.22 development cycle is yet to be confirmed.
+The plan for the 1.22 development cycle is yet to be confirmed. Assuming
+no major project-wide reorganisations in the 1.22 cycle we might try and
+aim for a release around August 2022.
1.22 will be backwards-compatible to the stable 1.20, 1.18, 1.16, 1.14,
1.12, 1.10, 1.8, 1.6, 1.4, 1.2 and 1.0 release series.
------------------------------------------------------------------------
These release notes have been prepared by Tim-Philipp Müller with
-contributions from …
+contributions from Matthew Waters, Nicolas Dufresne, Nirbheek Chauhan,
+Sebastian Dröge and Seungha Yang.
License: CC BY-SA 4.0
-This is GStreamer gst-plugins-base 1.19.3.
+This is GStreamer gst-plugins-base 1.19.90.
GStreamer 1.19 is the development branch leading up to the next major
stable version which will be 1.20.
<release>
<Version>
+ <revision>1.19.90</revision>
+ <branch>main</branch>
+ <name></name>
+ <created>2022-01-28</created>
+ <file-release rdf:resource="https://gstreamer.freedesktop.org/src/gst-plugins-base/gst-plugins-base-1.19.90.tar.xz" />
+ </Version>
+ </release>
+
+ <release>
+ <Version>
<revision>1.19.3</revision>
<branch>main</branch>
<name></name>
project('gst-plugins-base', 'c',
- version : '1.19.3.1',
+ version : '1.19.90',
meson_version : '>= 0.59',
default_options : [ 'warning_level=1',
'buildtype=debugoptimized' ])
+=== release 1.19.90 ===
+
+2022-01-28 14:28:35 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * NEWS:
+ * RELEASE:
+ * docs/gst_plugins_cache.json:
+ * gst-plugins-good.doap:
+ * meson.build:
+ Release 1.19.90
+
+2022-01-28 14:28:28 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * ChangeLog:
+ Update ChangeLogs for 1.19.90
+
2022-01-28 15:47:44 +0200 Sebastian Dröge <sebastian@centricular.com>
* ext/soup/gstsouphttpclientsink.c:
GStreamer 1.20 Release Notes
GStreamer 1.20 has not been released yet. It is scheduled for release in
-November 2021.
+late January / early February 2022.
1.19.x is the unstable development version that is being developed in
-the git main branch and which will eventually result in 1.20, and 1.19.3
-is the current development release in that series
-
-It is expected that feature freeze will be in early October 2021,
-followed by one or two 1.19.9x pre-releases and the new 1.20 stable
-release around October/November 2021.
+the git main branch and which will eventually result in 1.20, and
+1.19.90 is the first release candidate in that series (1.20rc1).
1.20 will be backwards-compatible to the stable 1.18, 1.16, 1.14, 1.12,
-1.10, 1.8, 1.6,, 1.4, 1.2 and 1.0 release series.
+1.10, 1.8, 1.6, 1.4, 1.2 and 1.0 release series.
See https://gstreamer.freedesktop.org/releases/1.20/ for the latest
version of this document.
-Last updated: Monday 1 November 2021, 01:00 UTC (log)
+Last updated: Wednesday 26 January 2022, 01:00 UTC (log)
Introduction
Highlights
-- this section will be completed in due course
+- Development in GitLab was switched to a single git repository
+ containing all the modules
+- GstPlay: new high-level playback library, replaces GstPlayer
+- WebM Alpha decoding support
+- Encoding profiles can now be tweaked with additional
+ application-specified element properties
+- Compositor: multi-threaded video conversion and mixing
+- RTP header extensions: unified support in RTP depayloader and
+ payloader base classes
+- SMPTE 2022-1 2-D Forward Error Correction support
+- Smart encoding (passthrough) support for VP8, VP9, H.265 in
+ encodebin and transcodebin
+- Runtime compatibility support for libsoup2 and libsoup3 (libsoup3
+ support experimental)
+- Video decoder subframe support
+- Video decoder automatic packet-loss, data corruption, and keyframe
+ request handling for RTP / WebRTC / RTSP
+- MP4 and Matroska muxers now support profile/level/resolution changes
+ for H264/H265 input streams (i.e. codec data changing on the fly)
+- MP4 muxing mode that initially creates a fragmented mp4 which is
+ converted to a regular mp4 on EOS
+- Audio support for the WebKit Port for Embedded (WPE) web page source
+ element
+- CUDA based video color space convert and rescale elements and
+ upload/download elements
+- NVIDIA memory:NVMM support for OpenGL glupload and gldownload
+ elements
+- Many WebRTC improvements
+- The new VA-API plugin implemention fleshed out with more decoders
+ and new postproc elements
+- AppSink API to retrieve events in addition to buffers and buffer
+ lists
+- AppSrc gained more configuration options for the internal queue
+ (leakiness, limits in buffers and time, getters to read current
+ levels)
+- Updated Rust bindings and many new Rust plugins
+- Improved support for custom minimal GStreamer builds
+- Support build against FFmpeg 5.0
+- Linux Stateless CODEC support gained MPEG2 and VP9
+- Windows Direct3D11/DXVA decoder gained AV1 and MPEG2 support
+- Lots of new plugins, features, performance improvements and bug
+ fixes
Major new features and changes
Noteworthy new features and API
-- this section will be filled in in due course
+- gst_element_get_request_pad() has been deprecated in favour of the
+ newly-added gst_element_request_pad_simple() which does the exact
+ same thing but has a less confusing name that hopefully makes clear
+ that the function request a new pad rather than just retrieves an
+ already-existing request pad.
+
+Development in GitLab was switched to a single git repository containing all the modules
+
+The GStreamer multimedia framework is a set of libraries and plugins
+split into a number of distinct modules which are released independently
+and which have so far been developed in separate git repositories in
+freedesktop.org GitLab.
+
+In addition to these separate git repositories there was a gst-build
+module that would use the Meson build systems’s subproject feature to
+download each individual module and then build everything in one go. It
+would also provide an uninstalled development environment that made it
+easy to work on GStreamer and use or test versions other than the
+system-installed GStreamer version.
+
+All of these modules have now (as of 28 September 2021) been merged into
+a single git repository (“Mono repository” or “monorepo”) which should
+simplify development workflows and continuous integration, especially
+where changes need to be made to multiple modules at once.
+
+This mono repository merge will primarily affect GStreamer developers
+and contributors and anyone who has workflows based on the GStreamer git
+repositories.
+
+The Rust bindings and Rust plugins modules have not been merged into the
+mono repository at this time because they follow a different release
+cycle.
+
+The mono repository lives in the existing GStreamer core git repository
+in GitLab in the new main branch and all future development will happen
+on this branch.
+
+Modules will continue to be released as separate tarballs.
+
+For more details, please see the GStreamer mono repository FAQ.
+
+GstPlay: new high-level playback library replacing GstPlayer
+
+- GstPlay is a new high-level playback library that replaces the older
+ GstPlayer API. It is basically the same API as GstPlayer but
+ refactored to use bus messages for application notifications instead
+ of GObject signals. There is still a signal adapter object for those
+ who prefer signals. Since the existing GstPlayer API is already in
+ use in various applications, it didn’t seem like a good idea to
+ break it entirely. Instead a new API was added, and it is expected
+ that this new GstPlay API will be moved to gst-plugins-base in
+ future.
+
+- The existing GstPlayer API is scheduled for deprecation and will be
+ removed at some point in the future (e.g. in GStreamer 1.24), so
+ application developers are urged to migrate to the new GstPlay API
+ at their earliest convenience.
+
+WebM alpha decoding
+
+- Implement WebM alpha decoding (VP8/VP9 with alpha), which required
+ support and additions in various places. This is supported both with
+ software decoders and hardware-accelerated decoders.
+
+- VP8/VP9 don’t support alpha components natively in the codec, so the
+ way this is implemented in WebM is by encoding the alpha plane with
+ transparency data as a separate VP8/VP9 stream. Inside the WebM
+ container (a variant of Matroska) this is coded as a single video
+ track with the “normal” VP8/VP9 video data making up the main video
+ data and each frame of video having an encoded alpha frame attached
+ to it as extra data ("BlockAdditional").
+
+- matroskademux has been extended extract this per-frame alpha side
+ data and attach it in form of a GstVideoCodecAlphaMeta to the
+ regular video buffers. Note that this new meta is specific to this
+ VP8/VP9 alpha support and can’t be used to just add alpha support to
+ other codecs that don’t support it. Lastly, matroskademux also
+ advertises the fact that the streams contain alpha in the caps.
+
+- The new codecalpha plugin contains various bits of infrastructure to
+ support autoplugging and debugging:
+
+ - codecalphademux splits out the alpha stream from the metas on
+ the regular VP8/VP9 buffers
+ - alphacombine takes two decoded raw video streams (one alpha, one
+ the regular video) and combines it into a video stream with
+ alpha
+ - vp8alphadecodebin + vp9alphadecodebin are wrapper bins that use
+ the regular vp8dec and vp9dec software decoders to decode
+ regular and alpha streams and combine them again. To decodebin
+ these look like regular decoders which ju
+ - The V4L2 CODEC plugin has stateless VP8/VP9 decoders that can
+ decode both alpha and non-alpha stream with a single decoder
+ instance
+
+- A new AV12 video format was added which is basically NV12 with an
+ alpha plane, which is more convenient for many hardware-accelerated
+ decoders.
+
+- Watch Nicolas Dufresne’s LCA 2022 talk “Bringing WebM Alpha support
+ to GStreamer” for all the details and a demo.
+
+RTP Header Extensions Base Class and Automatic Header Extension Handling in RTP Payloaders and Depayloaders
+
+- RTP Header Extensions are specified in RFC 5285 and provide a way to
+ add small pieces of data to RTP packets in between the RTP header
+ and the RTP payload. This is often used for per-frame metadata,
+ extended timestamps or other application-specific extra data. There
+ are several commonly-used extensions specified in various RFCs, but
+ senders are free to put any kind of data in there, as long as sender
+ and receiver both know what that data is. Receivers that don’t know
+ about the header extensions will just skip the extra data without
+ ever looking at it. These header extensions can often be combined
+ with any kind of payload format, so may need to be supported by many
+ RTP payloader and depayloader elements.
+
+- Inserting and extracting RTP header extension data has so far been a
+ bit inconvenient in GStreamer: There are functions to add and
+ retrieve RTP header extension data from RTP packets, but nothing
+ works automatically, even for common extensions. People would have
+ to do the insertion/extraction either in custom elements
+ before/after the RTP payloader/depayloader, or inside pad probes,
+ which isn’t very nice.
+
+- This release adds various pieces of new infrastructure for generic
+ RTP header extension handling, as well as some implementations for
+ common extensions:
+
+ - GstRTPHeaderExtension is a new helper base class for reading and
+ writing RTP header extensions. Nominally this subclasses
+ GstElement, but only so these extensions are stored in the
+ registry where they can be looked up by URI or name. They don’t
+ have pads and don’t get added to the pipeline graph as an
+ element.
+
+ - "add-extension" and "clear-extension" action signals on RTP
+ payloaders and depayloaders for manual extension management
+
+ - The "request-extension" signal will be emitted if an extension
+ is encountered that requires explicit mapping by the application
+
+ - new "auto-header-extension" property on RTP payloaders and
+ depayloaders for automatic handling of known header extensions.
+ This is enabled by default. The extensions must be signalled via
+ caps / SDP.
+
+ - RTP header extension implementations:
+
+ - rtphdrextclientaudiolevel: Client-to-Mixer Audio Level
+ Indication (RFC 6464) (also see below)
+ - rtphdrextcolorspace: Color Space extension, extends RTP
+ packets with color space and high dynamic range (HDR)
+ information
+ - rtphdrexttwcc: Transport Wide Congestion Control support
+
+- gst_rtp_buffer_remove_extension_data() is a new helper function to
+ remove an RTP header extension from an RTP buffer
+
+- The existing gst_rtp_buffer_set_extension_data() now also supports
+ shrinking the extension data in size
+
+AppSink and AppSrc improvements
+
+- appsink: new API to pull events out of appsink in addition to
+ buffers and buffer lists.
+
+ There was previously no way for users to receive incoming events
+ from appsink properly serialised with the data flow, even if they
+ are serialised events. The reason for that is that the only way to
+ intercept events was via a pad probe on the appsink sink pad, but
+ there is also internal queuing inside of appsink, so it’s difficult
+ to ascertain the right order of everything in all cases.
+
+ There is now a new "new-serialized-event" signal which will be
+ emitted when there’s a new event pending (just like the existing
+ "new-sample" signal). The "emit-signals" property must be set to
+ TRUE in order to activate this (but it’s also fine to just pull from
+ the application thread without using the signals).
+
+ gst_app_sink_pull_object() and gst_app_sink_try_pull_object() can be
+ used to pull out either an event or a new sample carrying a buffer
+ or buffer list, whatever is next in the queue.
+
+ EOS events will be filtered and will not be returned. EOS handling
+ can be done the ususal way, same as with _pull_sample().
+
+- appsrc: allow configuration of internal queue limits in time and
+ buffers and add leaky mode.
+
+ There is internal queuing inside appsrc so the application thread
+ can push data into the element which will then be picked up by the
+ source element’s streaming thread and pushed into the pipeline from
+ that streaming thread. This queue is unlimited by default and until
+ now it was only possible to set a maximum size limit in bytes. When
+ that byte limit is reached, the pushing thread (application thread)
+ would be blocked until more space becomes available.
+
+ A limit in bytes is not particularly useful for many use cases, so
+ now it is possible to also configure limits in time and buffers
+ using the new "max-time" and "max-buffers" properties. Of course
+ there are also matching new read-only"current-level-buffers" and
+ "current-level-time properties" properties to query the current fill
+ level of the internal queue in time and buffers.
+
+ And as if that wasn’t enough the internal queue can also be
+ configured as leaky using the new "leaky-type" property. That way
+ when the queue is full the application thread won’t be blocked when
+ it tries to push in more data, but instead either the new buffer
+ will be dropped or the oldest data in the queue will be dropped.
+
+Better string serialization of nested GstCaps and GstStructures
+
+- New string serialisation format for structs and caps that can handle
+ nested structs and caps properly by using brackets to delimit nested
+ items (e.g. some-struct, some-field=[nested-struct, nested=true]).
+ Unlike the default format the new variant can also support more than
+ one level of nesting. For backwards-compatibility reasons the old
+ format is still output by default when serialising caps and structs
+ using the existing API. The new functions gst_caps_serialize() and
+ gst_structure_serialize() can be used to output strings in the new
+ format.
+
+Convenience API for custom GstMetas
+
+- New convenience API to register and create custom GstMetas:
+ gst_meta_register_custom() and gst_buffer_add_custom_meta(). Such
+ custom meta is backed by a GstStructure and does not require that
+ users of the API expose their GstMeta implementation as public API
+ for other components to make use of it. In addition, it provides a
+ simpler interface by ignoring the impl vs. api distinction that the
+ regular API exposes. This new API is meant to be the meta
+ counterpart to custom events and messages, and to be more convenient
+ than the lower-level API when the absolute best performance isn’t a
+ requirement. The reason it’s less performant than a “proper” meta is
+ that a proper meta is just a C struct in the end whereas this goes
+ through the GstStructure API which has a bit more overhead, which
+ for most scenarios is negligible however. This new API is useful for
+ experimentation or proprietary metas, but also has some limitations:
+ it can only be used if there’s a single producer of these metas;
+ it’s not allowed to register the same custom meta multiple times or
+ from multiple places.
+
+Additional Element Properties on Encoding Profiles
+
+- GstEncodingProfile: The new "element-properties" and
+ gst_encoding_profile_set_element_properties() API allows
+ applications to set additional element properties on encoding
+ profiles to configure muxers and encoders. So far the encoding
+ profile template was the only place where this could be specified,
+ but often what applications want to do is take a ready-made encoding
+ profile shipped by GStreamer or the application and then tweak the
+ settings on top of that, which is now possible with this API. Since
+ applications can’t always know in advance what encoder element will
+ be used in the end, it’s even possible to specify properties on a
+ per-element basis.
+
+ Encoding Profiles are used in the encodebin, transcodebin and
+ camerabin elements and APIs to configure output formats (containers
+ and elementary streams).
+
+Audio Level Indication Meta for RFC 6464
+
+- New GstAudioLevelMeta containing Audio Level Indication as per RFC
+ 6464
+
+- The level element has been updated to add GstAudioLevelMeta on
+ buffers if the "audio-level-meta" property is set to TRUE. This can
+ then in turn be picked up by RTP payloaders to signal the audio
+ level to receivers through RTP header extensions (see above).
+
+- New Client-to-Mixer Audio Level Indication (RFC6464) RTP Header
+ Extension which should be automatically created and used by RTP
+ payloaders and depayloaders if their "auto-header-extension"
+ property is enabled and if the extension is part of the RTP caps.
+
+Automatic packet loss, data corruption and keyframe request handling for video decoders
+
+- The GstVideoDecoder base class has gained various new APIs to
+ automatically handle packet loss and data corruption better by
+ default, especially in RTP, RTSP and WebRTC streaming scenarios, and
+ to give subclasses more control about how they want to handle
+ missing data:
+
+ - Video decoder subclasses can mark output frames as corrupted via
+ the new GST_VIDEO_CODEC_FRAME_FLAG_CORRUPTED flag
+
+ - A new "discard-corrupted-frames" property allows applications to
+ configure decoders so that corrupted frames are directly
+ discarded instead of being forwarded inside the pipeline. This
+ is a replacement for the "output-corrupt" property of the FFmpeg
+ decoders.
+
+ - RTP depayloaders can now signal to decoders that data is missing
+ when sending GAP events for lost packets. GAP events can be sent
+ for various reason in a GStreamer pipeline. Often they are just
+ used to let downstream elements know that there isn’t a buffer
+ available at the moment, so downstream elements can move on
+ instead of waiting for one. They are also sent by RTP
+ depayloaders in the case that packets are missing, however, and
+ so far a decoder was not able to differentiate the two cases.
+ This has been remedied now: GAP events can be decorated with
+ gst_event_set_gap_flags() and GST_GAP_FLAG_MISSING_DATA to let
+ decoders now what happened, and decoders can then use that in
+ some cases to handle missing data better.
+
+ - The GstVideoDecoder::handle_missing_data vfunc was added to
+ inform subclasses about packet loss or missing data and let them
+ handle it in their own way if they like.
+
+ - gst_video_decoder_set_needs_sync_point() lets subclasses signal
+ that they need the stream to start with a sync point. If
+ enabled, the base class will discard all non-sync point frames
+ in the beginning and after a flush and does not pass them to the
+ subclass. Furthermore, if the first frame is not a sync point,
+ the base class will try and request a sync frame from upstream
+ by sending a force-key-unit event (see next items).
+
+ - New "automatic-request-sync-points" and
+ "automatic-request-sync-point-flags" properties to automatically
+ request sync points when needed, e.g. on packet loss or if the
+ first frame is not a keyframe. Applications may want to enable
+ this on decoders operating in e.g. RTP/WebRTC/RTSP receiver
+ pipelines.
+
+ - The new "min-force-key-unit-interval" property can be used to
+ ensure there’s a minimal interval between keyframe requests to
+ upstream (and/or the sender) and we’re not flooding the sender
+ with key unit requests.
+
+ - gst_video_decoder_request_sync_point() allows subclasses to
+ request a new sync point (e.g. if they choose to do their own
+ missing data handling). This will still honour the
+ "min-force-key-unit-interval" property if set.
+
+Improved support for custom minimal GStreamer builds
+
+- Element registration and registration of other plugin features
+ inside plugin init functions has been improved in order to
+ facilitate minimal custom GStreamer builds.
+
+- A number of new macros have been added to declare and create
+ per-element and per-pluginfeature register functions in all plugins,
+ and then call those from the per-plugin plugin_init functions:
+
+ - GST_ELEMENT_REGISTER_DEFINE,
+ GST_DEVICE_PROVIDER_REGISTER_DEFINE,
+ GST_DYNAMIC_TYPE_REGISTER_DEFINE, GST_TYPE_FIND_REGISTER_DEFINE
+ for the actual registration call with GStreamer
+ - GST_ELEMENT_REGISTER, GST_DEVICE_PROVIDER_REGISTER,
+ GST_DYNAMIC_TYPE_REGISTER, GST_PLUGIN_STATIC_REGISTER,
+ GST_TYPE_FIND_REGISTER to call the registration function defined
+ by the REGISTER_DEFINE macro
+ - GST_ELEMENT_REGISTER_DECLARE,
+ GST_DEVICE_PROVIDER_REGISTER_DECLARE,
+ GST_DYNAMIC_TYPE_REGISTER_DECLARE,
+ GST_TYPE_FIND_REGISTER_DECLARE to declare the registration
+ function defined by the REGISTER_DEFINE macro
+ - and various variants for advanced use cases.
+
+- This means that applications can call the per-element and
+ per-pluginfeature registration functions for only the elements they
+ need instead of registering plugins as a whole with all kinds of
+ elements that may not be required (e.g. encoder and decoder instead
+ of just decoder). In case of static linking all unused functions and
+ their dependencies would be removed in this case by the linker,
+ which helps minimise binary size for custom builds.
+
+- gst_init() will automatically call a gst_init_static_plugins()
+ function if one exists.
+
+- See the GStreamer static build documentation and Stéphane’s blog
+ post Generate a minimal GStreamer build, tailored to your needs for
+ more details.
New elements
-- this section will be filled in in due course
+- New aesdec and aesenc elements for AES encryption and decryption in
+ a custom format.
+
+- New encodebin2 element with dynamic/sometimes source pads in order
+ to support the option of doing the muxing outside of encodebin,
+ e.g. in combination with a splitmuxsink.
+
+- New fakeaudiosink and videocodectestsink elements for testing and
+ debugging (see below for more details)
+
+- rtpisacpay, rtpisacdepay: new RTP payloader and depayloader for iSAC
+ audio codec
+
+- rtpst2022-1-fecdec, rtpst2022-1-fecenc: new elements providing SMPTE
+ 2022-1 2-D Forward Error Correction. More details in Mathieu’s blog
+ post.
+
+- isac: new plugin wrapping the Internet Speech Audio Codec reference
+ encoder and decoder from the WebRTC project.
+
+- asio: plugin for Steinberg ASIO (Audio Streaming Input/Output) API
+
+- gssrc, gssink: add source and sink for Google Cloud Storage
+
+- onnx: new plugin to apply ONNX neural network models to video
+
+- openaptx: aptX and aptX-HD codecs using libopenaptx (v0.2.0)
+
+- qroverlay, debugqroverlay: new elements that allows overlaying data
+ on top of video in form of a QR code
+
+- cvtracker: new OpenCV-based tracker element
+
+- av1parse, vp9parse: new parsers for AV1 and VP9 video
+
+- va: work on the new VA-API plugin implementation for
+ hardware-accelerated video decoding and encoding has continued at
+ pace, with various new decoders and filters having joined the
+ initial vah264dec:
+
+ - vah265dec: VA-API H.265 decoder
+ - vavp8dec: VA-API VP8 decoder
+ - vavp9dec: VA-API VP9 decoder
+ - vaav1dec: VA-API AV1 decoder
+ - vampeg2dec: VA-API MPEG-2 decoder
+ - vadeinterlace: : VA-API deinterlace filter
+ - vapostproc: : VA-API postproc filter (color conversion,
+ resizing, cropping, color balance, video rotation, skin tone
+ enhancement, denoise, sharpen)
+
+ See Víctor’s blog post “GstVA in GStreamer 1.20” for more details
+ and what’s coming up next.
+
+- vaapiav1dec: new AV1 decoder element (in gstreamer-vaapi)
+
+- msdkav1dec: hardware-accelerated AV1 decoder using the Intel Media
+ SDK / oneVPL
+
+- nvcodec plugin for NVIDIA NVCODEC API for hardware-accelerated video
+ encoding and decoding:
+
+ - cudaconvert, cudascale: new CUDA based video color space convert
+ and rescale elements
+ - cudaupload, cudadownload: new helper elements for memory
+ transfer between CUDA and system memory spaces
+ - nvvp8sldec, nvvp9sldec: new GstCodecs-based VP8/VP9 decoders
+
+- Various new hardware-accelerated elements for Windows:
+
+ - d3d11screencapturesrc: new desktop capture element, including a
+ GstDeviceProvider implementation to enumerate/select target
+ monitors for capture.
+ - d3d11av1dec and d3d11mpeg2dec: AV1 and MPEG-2 decoders
+ - d3d11deinterlace: deinterlacing filter
+ - d3d11compositor: video composing element
+ - see Windows section below for more details
+
+- new Rust plugins:
+
+ - audiornnoise: Removes noise from an audio stream
+ - awstranscribeparse: Parses AWS audio transcripts into timed text
+ buffers
+ - ccdetect: Detects if valid closed captions are present in a
+ closed captions stream
+ - cea608tojson: Converts CEA-608 Closed Captions to a JSON
+ representation
+ - cmafmux: CMAF fragmented MP4 muxer
+ - dashmp4mux: DASH fragmented MP4 muxer
+ - isofmp4mux: ISO fragmented MP4 muxer
+ - ebur128level: EBU R128 Loudness Level Measurement
+ - ffv1dec: FFV1 video decoder
+ - gtk4paintablesink: GTK4 video sink, which provides a
+ GdkPaintable that can be rendered in various widgets
+ - hlssink3: HTTP Live Streaming sink
+ - hrtfrender: Head-Related Transfer Function (HRTF) renderer
+ - hsvdetector: HSV colorspace detector
+ - hsvfilter: HSV colorspace filter
+ - jsongstenc: Wraps buffers containing any valid top-level JSON
+ structures into higher level JSON objects, and outputs those as
+ ndjson
+ - jsongstparse: Parses ndjson as output by jsongstenc
+ - jsontovtt: converts JSON to WebVTT subtitles
+ - regex: Applies regular expression operations on text
+ - roundedcorners: Adds rounded corners to video
+ - spotifyaudiosrc: Spotify source
+ - textahead: Display upcoming text buffers ahead (e.g. for
+ Karaoke)
+ - transcriberbin: passthrough bin that transcribes raw audio to
+ closed captions using awstranscriber and puts the captions as
+ metas onto the video
+ - tttojson: Converts timed text to a JSON representation
+ - uriplaylistbin: Playlist source bin
+ - webpdec-rs: WebP image decoder with animation support
+
+- New plugin codecalpha with elements to assist with WebM Alpha
+ decoding
+
+ - codecalphademux: Split stream with GstVideoCodecAlphaMeta into
+ two streams
+ - alphacombine: Combine two raw video stream (I420 or NV12) as one
+ stream with alpha channel (A420 or AV12)
+ - vp8alphadecodebin: A bin to handle software decoding of VP8 with
+ alpha
+ - vp9alphadecodebin: A bin to handle software decoding of VP9 with
+ alpha
+
+- New hardware accelerated elements for Linux:
+
+ - v4l2slmpeg2dec: Support for Linux Stateless MPEG2 decoders
+ - v4l2slvp9dec: Support for Linux Stateless VP9 decoders
+ - v4l2slvp8alphadecodebin: Support HW accelerated VP8 with alpha
+ layer decoding
+ - v4l2slvp9alphadecodebin: Support HW accelerated VP9 with alpha
+ layer decoding
New element features and additions
-- this section will be filled in in due course
+- assrender: handle more font mime types; better interaction with
+ matroskademux for embedded fonts
+
+- audiobuffersplit: Add support for specifying output buffer size in
+ bytes (not just duration)
+
+- audiolatency: new "samplesperbuffer" property so users can configure
+ the number of samples per buffer. The default value is 240 samples
+ which is equivalent to 5ms latency with a sample rate of 48000,
+ which might be larger than actual buffer size of audio capture
+ device.
+
+- audiomixer, audiointerleave, GstAudioAggregator: now keep a count of
+ samples that are dropped or processed as statistic and can be made
+ to post QoS messages on the bus whenever samples are dropped by
+ setting the "qos-messages" property on input pads.
+
+- audiomixer, compositor: improved handling of new inputs added at
+ runtime. New API was added to the GstAggregator base class to allow
+ subclasses to opt into an aggregation mode where inactive pads are
+ ignored when processing input buffers
+ (gst_aggregator_set_ignore_inactive_pads(),
+ gst_aggregator_pad_is_inactive()). An “inactive pad” in this context
+ is a pad which, in live mode, hasn’t yet received a first buffer,
+ but has been waited on at least once. What would happen usually in
+ this case is that the aggregator would wait for data on this pad
+ every time, up to the maximum configured latency. This would
+ inadvertently push mixer elements in live mode to the configured
+ latency envelope and delay processing when new inputs are added at
+ runtime until these inputs have actually produced data. This is
+ usually undesirable. With this new API, new inputs can be added
+ (requested) and configured and they won’t delay the data processing.
+ Applications can opt into this new behaviour by setting the
+ "ignore-inactive-pads" property on compositor, audiomixer or other
+ GstAudioAggregator-based elements.
+
+- cccombiner: implement “scheduling” of captions. So far cccombiner’s
+ behaviour was essentially that of a funnel: it strictly looked at
+ input timestamps to associate together video and caption buffers.
+ Now it will try to smoothly schedule caption buffers in order to
+ have exactly one per output video buffer. This might involve
+ rewriting input captions, for example when the input is CDP then
+ sequence counters are rewritten, time codes are dropped and
+ potentially re-injected if the input video frame had a time code
+ meta. This can also lead to the input drifting from synchronization,
+ when there isn’t enough padding in the input stream to catch up. In
+ that case the element will start dropping old caption buffers once
+ the number of buffers in its internal queue reaches a certain limit
+ (configurable via the "max-scheduled" property). The new original
+ funnel-like behaviour can be restored by setting the "scheduling"
+ property to FALSE.
+
+- ccconverter: new "cdp-mode" property to specify which sections to
+ include in CDP packets (timecode, CC data, service info). Various
+ software, including ffmpeg’s Decklink support, fails parsing CDP
+ packets that contain anything but CC data in the CDP packets.
+
+- clocksync: new "sync-to-first" property for automatic timestamp
+ offset setup: if set clocksync will set up the "ts-offset" value
+ based on the first buffer and the pipeline’s running time when the
+ first buffer arrived. The newly configured "ts-offset" in this case
+ would be the value that allows outputting the first buffer without
+ waiting on the clock. This is useful for example to feed a non-live
+ input into an already-running pipeline.
+
+- compositor:
+
+ - multi-threaded input conversion and compositing. Set the
+ "max-threads" property to activate this.
+ - new "sizing-policy" property to support display aspect ratio
+ (DAR)-aware scaling. By default the image is scaled to fill the
+ configured destination rectangle without padding and without
+ keeping the aspect ratio. With sizing-policy=keep-aspect-ratio
+ the input image is scaled to fit the destination rectangle
+ specified by GstCompositorPad:{xpos, ypos, width, height}
+ properties preserving the aspect ratio. As a result, the image
+ will be centered in the destination rectangle with padding if
+ necessary.
+ - new "zero-size-is-unscaled" property on input pads. By default
+ pad width=0 or pad height=0 mean that the stream should not be
+ scaled in that dimension. But if the "zero-size-is-unscaled"
+ property is set to FALSE a width or height of 0 is instead
+ interpreted to mean that the input image on that pad should not
+ be composited, which is useful when creating animations where an
+ input image is made smaller and smaller until it disappears.
+ - improved handling of new inputs at runtime via
+ "ignore-inactive-pads"property (see above for details)
+ - allow output format with alpha even if none of the inputs have
+ alpha (also glvideomixer and other GstVideoAggregator
+ subclasses)
+
+- dashsink: add h265 codec support and signals for allowing custom
+ playlist/fragment output
+
+- decodebin3:
+
+ - improved decoder selection, especially for hardware decoders
+ - make input activation “atomic” when adding inputs dynamically
+ - better interleave handling: take into account decoder latency
+ for interleave size
+
+- decklink:
+
+ - Updated DeckLink SDK to 11.2 to support DeckLink 8K Pro
+ - decklinkvideosrc:
+ - More accurate and stable capture timestamps: use the
+ hardware reference clock time when the frame was finished
+ being captured instead of a clock time much further down the
+ road.
+ - Automatically detect widescreen vs. normal NTSC/PAL
+
+- encodebin:
+
+ - add “smart encoding” support for H.265, VP8 and VP9 (i.e. only
+ re-encode where needed and otherwise pass through encoded video
+ as-is).
+ - H264/H265 smart encoding improvements: respect user-specified
+ stream-format, but if not specified default to avc3/hvc1 with
+ in-band SPS/PPS/VPS signalling for more flexibility.
+ - new encodebin2 element with dynamic/sometimes source pads in
+ order to support the option of doing the muxing outside of
+ encodebin, e.g. in combination with splitmuxsink.
+ - add APIs to set element properties on encoding profiles (see
+ below)
+
+- errorignore: new "ignore-eos" property to also ignore FLOW_EOS from
+ downstream elements
+
+- giosrc: add support for growing source files: applications can
+ specify that the underlying file being read is growing by setting
+ the "is-growing" property. If set, the source won’t EOS when it
+ reaches the end of the file, but will instead start monitoring it
+ and will start reading data again whenever a change is detected. The
+ new "waiting-data" and "done-waiting-data" signals keep the
+ application informed about the current state.
+
+- gtksink, gtkglsink:
+
+ - scroll event support: forwarded as navigation events into the
+ pipeline
+ - "video-aspect-ratio-override" property to force a specific
+ aspect ratio
+ - "rotate-method" property and support automatic rotation based on
+ image tags
+
+- identity: new "stats" property allows applications to retrieve the
+ number of bytes and buffers that have passed through so far.
+
+- interlace: add support for more formats, esp 10-bit, 12-bit and
+ 16-bit ones
+
+- jack: new "low-latency" property for automatic latency-optimized
+ setting and "port-names" property to select ports explicitly
+
+- jpegdec: support output conversion to RGB using libjpeg-turbo (for
+ certain input files)
+
+- line21dec:
+
+ - "mode" property to control whether and how detected closed
+ captions should be inserted in the list of existing close
+ caption metas on the input frame (if any): add, drop, or
+ replace.
+ - "ntsc-only" property to only look for captions if video has NTSC
+ resolution
+
+- line21enc: new "remove-caption-meta" to remove metas from output
+ buffers after encoding the captions into the video data; support for
+ CDP closed captions
+
+- matroskademux, matroskamux: Add support for ffv1, a lossless
+ intra-frame video coding format.
+
+- matroskamux: accept in-band SPS/PPS/VPS for H264 and H265
+ (i.e. stream-format avc3 and hev1) which allows on-the-fly
+ profile/level/resolution changes.
+
+- matroskamux: new "cluster-timestamp-offset" property, useful for use
+ cases where the container timestamps should map to some absolute
+ wall clock time, for example.
+
+- rtpsrc: add "caps" property to allow explicit setting of the caps
+ where needed
+
+- mpegts: support SCTE-35 passthrough via new "send-scte35-events"
+ property on MPEG-TS demuxer tsdemux. When enabled, SCTE 35 sections
+ (eg ad placement opportunities) are forwarded as events donwstream
+ where they can be picked up again by mpegtsmux. This required a
+ semantic change in the SCTE-35 section API: timestamps are now in
+ running time instead of muxer pts.
+
+- tsdemux: Handle PCR-less MPEG-TS streams; more robust timestamp
+ handling in certain corner cases and for poorly muxed streams.
+
+- mpegtsmux:
+
+ - More conformance improvements to make MPEG-TS analyzers happy:
+ - PCR timing accuracy: Improvements to the way mpegtsmux
+ outputs PCR observations in CBR mode, so that a PCR
+ observation is always inserted when needed, so that we never
+ miss the configured pcr-interval, as that triggers various
+ MPEG-TS analyser errors.
+ - Improved PCR/SI scheduling
+ - Don’t write PCR until PAT/PMT are output to make sure streams
+ start cleanly with a PAT/PMT.
+ - Allow overriding the automatic PMT PID selection via
+ application-supplied PMT_%d fields in the prog-map
+ structure/property.
+
+- mp4mux:
+
+ - new "first-moov-then-finalise" mode for fragmented output where
+ the output will start with a self-contained moov atom for the
+ first fragment, and then produce regular fragments. Then at the
+ end when the file is finalised, the initial moov is invalidated
+ and a new moov is written covering the entire file. This way the
+ file is a “fragmented mp4” file while it is still being written
+ out, and remains playable at all times, but at the end it is
+ turned into a regular mp4 file (with former fragment headers
+ remaining as unused junk data in the file).
+ - support H.264 avc3 and H.265 hvc1 stream formats as input where
+ the codec data is signalled in-band inside the bitstream instead
+ of caps/file headers.
+ - support profile/level/resolution changes for H264/H265 input
+ streams (i.e. codec data changing on the fly). Each codec_data
+ is put into its own SampleTableEntry inside the stsd, unless the
+ input is in avc3 stream format in which case it’s written
+ in-band an not in the headers.
+
+- multifilesink: new ""min-keyframe-distance"" property to make
+ minimum distance between keyframes in next-file=key-frame mode
+ configurable instead of hard-coding it to 10 seconds.
+
+- mxfdemux has seen a big refactoring to support non-frame wrappings
+ and more accurate timestamp/seek handling for some formats
+
+- msdk plugin for hardware-accelerated video encoding and decoding
+ using the Intel Media SDK:
+
+ - oneVPL support (Intel oneAPI Video Processing Library)
+ - AV1 decoding support
+ - H.264 decoder now supports constrained-high and progressive-high
+ profiles
+ - H.264 encoder:
+ - more configuration options (properties):
+ "intra-refresh-type", "min-qp" , "max-qp", "p-pyramid",
+ "dblk-idc"
+ - H.265 encoder:
+ - can output main-still-picture profile
+ - now inserts HDR SEIs (mastering display colour volume and
+ content light level)
+ - more configuration options (properties):
+ "intra-refresh-type", "min-qp" , "max-qp", "p-pyramid",
+ "b-pyramid", "dblk-idc", "transform-skip"
+ - support for RGB 10bit format
+ - External bitrate control in encoders
+ - Video post proc element msdkvpp gained support for 12-bit pixel
+ formats P012_LE, Y212_LE and Y412_LE
+
+- nvh264sldec: interlaced stream support
+
+- openh264enc: support main, high, constrained-high and
+ progressive-high profiles
+
+- openjpeg: support for multithreaded decoding and encoding
+
+- rtspsrc: now supports IPv6 also for tunneled mode (RTSP-over-HTTP);
+ new "ignore-x-server-reply" property to ignore the
+ x-server-ip-address server header reply in case of HTTP tunneling,
+ as it is often broken.
+
+- souphttpsrc: Runtime compatibility support for libsoup2 and
+ libsoup3. libsoup3 is the latest major version of libsoup, but
+ libsoup2 and libsoup3 can’t co-exist in the same process because
+ there is no namespacing or versioning for GObject types. As a
+ result, it would be awkward if the GStreamer souphttpsrc plugin
+ linked to a specific version of libsoup, because it would only work
+ with applications that use the same version of libsoup. To make this
+ work, the soup plugin now tries to determine the libsoup version
+ used by the application (and its other dependencies) at runtime on
+ systems where GStreamer is linked dynamically. libsoup3 support is
+ still considered somewhat experimental at this point.
+
+- srtsrc, srtsink: add signals for the application to accept/reject
+ incoming connections
+
+- timeoverlay: new elapsed-running-time time mode which shows the
+ running time since the first running time (and each flush-stop).
+
+- udpsrc: new timestamping mode to retrieve packet receive timestamps
+ from the kernel via socket control messages (SO_TIMESTAMPNS) on
+ supported platforms
+
+- uritranscodebin: new setup-source and element-setup signals for
+ applications to configure elements used
+
+- v4l2codecs plugin gained support for 4x4 and 32x32 tile formats
+ enabling some platforms or direct renders. Important memory usage
+ improvement.
+
+- v4l2slh264dec now implements the final Linux uAPI as shipped on
+ Linux 5.11 and later.
+
+- valve: add "drop-mode" property and provide two new modes of
+ operation: in drop-mode=forward-sticky-events sticky events
+ (stream-start, segment, tags, caps, etc.) are forwarded downstream
+ even when dropping is enabled; drop-mode=transform-to-gap will in
+ addition also convert buffers into gap events when dropping is
+ enabled, which lets downstream elements know that time is advancing
+ and might allow for preroll in many scenarios. By default all events
+ and all buffers are dropped when dropping is enabled, which can
+ cause problems with caps negotiation not progressing or branches not
+ prerolling when dropping is enabled.
+
+- videocrop: support for many more pixel formats, e.g. planar YUV
+ formats with > 8bits and GBR* video formats; can now also accept
+ video not backed by system memory as long as downstream supports the
+ GstCropMeta
+
+- videotestsrc: new smpte-rp-219 pattern for SMPTE75 RP-219 conformant
+ color bars
+
+- vp8enc: finish support for temporal scalability: two new properties
+ ("temporal-scalability-layer-flags",
+ "temporal-scalability-layer-sync-flags") and a unit change on the
+ "temporal-scalability-target-bitrate" property (now expects bps);
+ also make temporal scalability details available to RTP payloaders
+ as buffer metadata.
+
+- vp9enc: new properties to tweak encoder performance:
+
+ - "aq-mode" to configure adaptive quantization modes
+ - "frame-parallel-decoding" to configure whether to create a
+ bitstream that reduces decoding dependencies between frames
+ which allows staged parallel processing of more than one video
+ frames in the decoder. (Defaults to TRUE)
+ - "row-mt", "tile-columns" and "tile-rows" so multithreading can
+ be enabled on a per-tile basis, instead of on a per tile-column
+ basis. In combination with the new "tile-rows" property, this
+ allows the encoder to make much better use of the available CPU
+ power.
+
+- vp9dec, vp9enc: add support for 10-bit 4:2:0 and 4:2:2 YUV, as well
+ as 8-bit 4:4:4
+
+- vp8enc, vp9enc now default to “good quality” for the deadline
+ property rather then “best quality”. Having the deadline set to best
+ quality causes the encoder to be absurdly slow, most real-life users
+ will prefer good-enough quality with better performance instead.
+
+- wpesrc:
+
+ - implement audio support: a new sometimes source pad will be
+ created for each audio stream created by the web engine.
+ - move wpesrc to wpevideosrc and add a wrapper bin wpesrc to also
+ support audio
+ - also handles web:// URIs now (same as cefsrc)
+ - post messages with the estimated load progress on the bus
-Plugin and library moves
+- x265enc: add negative DTS support, which means timestamps are now
+ offset by 1h same as with x264enc
+
+RTP Payloaders and Depayloaders
+
+- rtpisacpay, rtpisacdepay: new RTP payloader and depayloader for iSAC
+ audio codec
+
+- rtph264depay:
+
+ - new "request-keyframe" property to make the depayloader
+ automatically request a new keyframe from the sender on packet
+ loss, consistent with the new property on rtpvp8depay.
+ - new "wait-for-keyframe" property to make depayloader wait for a
+ new keyframe at the beginning and after packet loss (only
+ effective if the depayloader outputs AUs), consistent with the
+ existing property on rtpvp8depay.
+
+- rtpopuspay, rtpopusdepay: support libwebrtc-compatible multichannel
+ audio in addition to the previously supported multichannel audio
+ modes
+
+- rtpopuspay: add DTX (Discontinuous Transmission) support
+
+- rtpvp8depay: new "request-keyframe" property to make the depayloader
+ automatically request a new keyframe from the sender on packet loss.
+
+- rtpvp8pay: temporal scaling support
-- this section will be filled in in due course
+- rtpvp9depay: Improved SVC handling (aggregate all layers)
+
+RTP Infrastructure
+
+- rtpst2022-1-fecdec, rtpst2022-1-fecenc: new elements providing SMPTE
+ 2022-1 2-D Forward Error Correction. More details in Mathieu’s blog
+ post.
+
+- rtpreddec: BUNDLE support
+
+- rtpredenc, rtpulpfecenc: add support for Transport-wide Congestion
+ Control (TWCC)
+
+- rtpsession: new "twcc-feedback-interval" property to allow RTCP TWCC
+ reports to be scheduled on a timer instead of per marker-bit.
+
+Plugin and library moves
- There were no plugin moves or library moves in this cycle.
The following elements or plugins have been removed:
-- this section will be filled in in due course
-
- The ofa audio fingerprinting plugin has been removed. The MusicIP
database has been defunct for years so this plugin is likely neither
useful nor used by anyone.
Miscellaneous API additions
-- this section will be filled in in due course
+Core
+
+- gst_buffer_new_memdup() is a convenience function for the
+ widely-used gst_buffer_new_wrapped(g_memdup(data,size),size)
+ pattern.
+
+- gst_caps_features_new_single() creates a new single GstCapsFeatures,
+ avoiding the need to use the vararg function with NULL terminator
+ for simple cases.
+
+- gst_element_type_set_skip_documentation() can be used by plugins to
+ signal that certain elements should not be included in the GStreamer
+ plugin documentation. This is useful for plugins where elements are
+ registered dynamically based on hardware capabilities and/or where
+ the available plugins and properties vary from system to system.
+ This is used in the d3d11 plugin for example to ensure that only the
+ list of default elements is advertised in the documentation.
+
+- gst_type_find_suggest_empty_simple() is a new convenience function
+ for typefinders for cases where there’s only a media type and no
+ other fields.
+
+- New API to create elements and set properties at construction time,
+ which is not only convenient, but also allows GStreamer elements to
+ have construct-only properties: gst_element_factory_make_full(),
+ gst_element_factory_make_valist(),
+ gst_element_factory_make_with_properties(),
+ gst_element_factory_create_full(),
+ gst_element_factory_create_valist(),
+ gst_element_factory_create_with_properties().
+
+- GstSharedTaskPool: new “shared” task pool subclass with slightly
+ different default behaviour than the existing GstTaskPool which
+ would create unlimited number of threads for new tasks. The shared
+ taskpool creates up to N threads (default: 1) and then distributes
+ pending tasks to those threads round-robin style, and blocks if no
+ thread is available. It is possible to join tasks. This can be used
+ by plugins to implement simple multi-threaded processing and is used
+ for the new multi-threaded video conversion and compositing done in
+ GstVideoAggregator, videoconverter and compositor.
+
+Plugins Base Utils library
+
+- GstDiscoverer:
+
+ - gst_discoverer_container_info_get_tags() was added to retrieve
+ global/container tags (vs. per-stream tags). Per-Stream tags can
+ be retrieved via the existing
+ gst_discoverer_stream_info_get_tags().
+ gst_discoverer_info_get_tags(), which for many files returns a
+ confusing mix of stream and container tags, has been deprecated
+ in favour of the container/stream-specific functions.
+ - gst_discoverer_stream_info_get_stream_number() returns a unique
+ integer identifier for a given stream within the given
+ GstDiscoverer context. (If this matches the stream number inside
+ the container bitstream that’s by coincidence and not by
+ design.)
+
+- gst_pb_utils_get_caps_description_flags() can be used to query
+ whether certain caps represent a container, audio, video, image,
+ subtitles, tags, or something else. This only works for formats
+ known to GStreamer.
+
+- gst_pb_utils_get_file_extension_from_caps() returns a possible file
+ extension for given caps.
+
+- gst_codec_utils_h264_get_profile_flags_level(): Parses profile,
+ flags, and level from H264 AvcC codec_data. The format of H264 AVCC
+ extradata/sequence_header is documented in the ITU-T H.264
+ specification section 7.3.2.1.1 as well as in ISO/IEC 14496-15
+ section 5.3.3.1.2.
+
+- gst_codec_utils_caps_get_mime_codec() to convert caps to a RFC 6381
+ compatible MIME codec string codec. Useful for providing the codecs
+ field inside the Content-Type HTTP header for containerized formats,
+ such as mp4 or matroska.
+
+GStreamer OpenGL integration library and plugins
+
+- glcolorconvert: added suppport for converting the video formats
+ A420, AV12, BGR, BGRA, RGBP and BGRP.
+
+- Added support to GstGLBuffer for persistent buffer mappings where a
+ Pixel Buffer Object (PBO) can be mapped by both the CPU and the GPU.
+ This removes a memcpy() when uploading textures or vertices
+ particularly when software decoders (e.g. libav) are direct
+ rendering into our memory. Improves transfer performance
+ significantly. Requires OpenGL 4.4, GL_ARB_buffer_storage or
+ GL_EXT_buffer_storage
+
+- Added various helper functions for handling 4x4 matrices of affine
+ transformations as used by GstVideoAffineTransformationMeta.
+
+- Add support to GstGLContext for allowing the application to control
+ the config (EGLConfig, GLXConfig, etc) used when creating the OpenGL
+ context. This allows the ability to choose between RGB16 or RGB10A2
+ or RGBA8 back/front buffer configurations that were previously
+ hardcoded. GstGLContext also supports retrieving the configuration
+ it was created with or from an externally provide OpenGL context
+ handle. This infrastructure is also used to create a compatible
+ config from an application/externally provided OpenGL context in
+ order to improve compatibility with other OpenGL frameworks and GUI
+ toolkits. A new environment variable GST_GL_CONFIG was also added to
+ be able to request a specific configuration from the command line.
+ Note: different platforms will have different functionality
+ available.
+
+- Add support for choosing between EGL and WGL at runtime when running
+ on Windows. Previously this was a build-time switch. Allows use in
+ e.g. Gtk applications on Windows that target EGL/ANGLE without
+ recompiling GStreamer. gst_gl_display_new_with_type() can be used by
+ applications to choose a specific display type to use.
+
+- Build fixes to explicitly check for Broadcom-specific libraries on
+ older versions of the Raspberry Pi platform. The Broadcom OpenGL ES
+ and EGL libraries have different filenames. Using the vc4 Mesa
+ driver on the Raspberry Pi is not affected.
+
+- Added support to glupload and gldownload for transferring RGBA
+ buffers using the memory:NVMM available on the Nvidia Tegra family
+ of embedded devices.
+
+- Added support for choosing libOpenGL and libGLX as used in a GLVND
+ environment on unix-based platforms. This allows using desktop
+ OpenGL and EGL without pulling in any GLX symbols as would be
+ required with libGL.
+
+Video library
+
+- New raw video formats:
+
+ - AV12 (NV12 with alpha plane)
+ - RGBP and BGRP (planar RGB formats)
+ - ARGB64 variants with specified endianness instead of host
+ endianness:
+ - ARGB64_LE, ARGB64_BE
+ - RGBA64_BE, RGBA64_LE
+ - BGRA64_BE, BGRA64_LE
+ - ABGR64_BE, ABGR64_LE
+
+- gst_video_orientation_from_tag() is new convenience API to parse the
+ image orientation from a GstTagList.
+
+- GstVideoDecoder subframe support (see below)
+
+- GstVideoCodecState now also carries some HDR metadata
+
+- Ancillary video data: implement transform functions for AFD/Bar
+ metas, so they will be forwarded in more cases
+
+MPEG-TS library
+
+This library only handles section parsing and such, see above for
+changes to the actual mpegtsmux and mpegtsdemux elements.
+
+- many additions and improvements to SCTE-35 section parsing
+- new API for fetching extended descriptors:
+ gst_mpegts_find_descriptor_with_extension()
+- add support for SIT sections (Selection Information Tables)
+- expose event-from-section constructor gst_event_new_mpegts_section()
+- parse Audio Preselection Descriptor needed for Dolby AC-4
+
+GstWebRTC library + webrtcbin
+
+- Change the way in which sink pads and transceivers are matched
+ together to support easier usage. If a pad is created without a
+ specific index (i.e. using sink_%u as the pad template), then an
+ available compatible transceiver will be searched for. If a specific
+ index is requested (i.e. sink_1) then if a transceiver for that
+ m-line already exists, that transceiver must match the new sink pad
+ request. If there is no transceiver available in either scenario, a
+ new transceiver is created. If a mixture of both sink_1 and sink_%u
+ requests result in an impossible situation, an error will be
+ produced at pad request time or from create offer/answer.
+
+- webrtcbin now uses regular ICE nomination instead of libnice’s
+ default of aggressive ICE nomination. Regular ICE nomination is the
+ default recommended by various relevant standards and improves
+ connectivity in specific network scenarios.
+
+- Add support for limiting the port range used for RTP with the
+ addition of the min-rtp-port and max-rtp-port properties on the ICE
+ object.
+
+- Expose the SCTP transport as a property on webrtcbin to more closely
+ match the WebRTC specification.
+
+- Added support for taking into account the data channel transport
+ state when determining the value of the "connection-state" property.
+ Previous versions of the WebRTC spec did not include the data
+ channel state when computing this value.
+
+- Add configuration for choosing the size of the underlying sockets
+ used for transporting media data
+
+- Always advertise support for the transport-cc RTCP feedback protocol
+ as rtpbin supports it. For full support, the configured caps (input
+ or through codec-preferences) need to include the relevant RTP
+ header extension.
+
+- Numerous fixes to caps and media handling to fail-fast when an
+ incompatible situation is detected.
+
+- Improved support for attaching the required media after a remote
+ offer has been set.
+
+- Add support for dynamically changing the amount of FEC used for a
+ particular stream.
+
+- webrtcbin now stops further SDP processing at the first error it
+ encounters.
+
+- Completed support for either local or the remote closing a data
+ channel.
+
+- Various fixes when performing BUNDLEing of the media streams in
+ relation to RTX and FEC usage.
+
+- Add support for writing out QoS DSCP marking on outgoing packets to
+ improve reliability in some network scenarios.
+
+- Improvements to the statistics returned by the get-stats signal
+ including the addition of the raw statistics from the internal
+ RTPSource, the TWCC stats when available.
+
+- The webrtc library does not expose any objects anymore with public
+ fields. Instead properties have been added to replace that
+ functionality. If you are accessing such fields in your application,
+ switch to the corresponding properties.
+
+GstCodecs and Video Parsers
+
+- Support for render delays to improve throughput across all CODECs
+ (used with NVDEC and V4L2).
+- lots of improvements to parsers and the codec parsing decoder base
+ classes (H264, H265, VP8, VP9, AV1, MPEG-2) used for various
+ hardware-accelerate decoder APIs.
+
+Bindings support
+
+- gst_allocation_params_new() allocates a GstAllocationParams struct
+ on the heap. This should only be used by bindings (and freed via
+ gst_allocation_params_free() then). In C code you would allocate
+ this on the stack and only init it in place.
+
+- gst_debug_log_literal() can be used to log a string to the debug log
+ without going through any printf format expansion and associated
+ overhead. This is mostly useful for bindings such as the Rust
+ bindings which may have done their own formatting already .
+
+- Provide non-inlined versions of refcounting APIs for various
+ GStreamer mini objects, so that they can be consumed by bindings
+ (e.g. gstreamer-sharp): gst_buffer_ref, gst_buffer_unref,
+ gst_clear_buffer, gst_buffer_copy, gst_buffer_replace,
+ gst_buffer_list_ref, gst_buffer_list_unref, gst_clear_buffer_list,
+ gst_buffer_list_copy, gst_buffer_list_replace, gst_buffer_list_take,
+ gst_caps_ref, gst_caps_unref, gst_clear_caps, gst_caps_replace,
+ gst_caps_take, gst_context_ref, gst_context_unref, gst_context_copy,
+ gst_context_replace, gst_event_replace, gst_event_steal,
+ gst_event_take, gst_event_ref, gst_event_unref, gst_clear_event,
+ gst_event_copy, gst_memory_ref, gst_memory_unref, gst_message_ref,
+ gst_message_unref, gst_clear_message, gst_message_copy,
+ gst_message_replace, gst_message_take, gst_promise_ref,
+ gst_promise_unref, gst_query_ref, gst_query_unref, gst_clear_query,
+ gst_query_copy, gst_query_replace, gst_query_take, gst_sample_ref,
+ gst_sample_unref, gst_sample_copy, gst_tag_list_ref,
+ gst_tag_list_unref, gst_clear_tag_list, gst_tag_list_replace,
+ gst_tag_list_take, gst_uri_copy, gst_uri_ref, gst_uri_unref,
+ gst_clear_uri.
+
+- expose a GType for GstMiniObject
+
+- gst_device_provider_probe() now returns non-floating device object
+
+API Deprecations
+
+- gst_element_get_request_pad() has been deprecated in favour of the
+ newly-added gst_element_request_pad_simple() which does the exact
+ same thing but has a less confusing name that hopefully makes clear
+ that the function request a new pad rather than just retrieves an
+ already-existing request pad.
+
+- gst_discoverer_info_get_tags(), which for many files returns a
+ confusing mix of stream and container tags, has been deprecated in
+ favour of the container-specific and stream-specific functions,
+ gst_discoverer_container_info_get_tags() and
+ gst_discoverer_stream_info_get_tags().
+
+- gst_video_sink_center_rect() was deprecated in favour of the more
+ generic newly-added gst_video_center_rect().
+
+- The GST_MEMORY_FLAG_NO_SHARE flag has been deprecated, as it tends
+ to cause problems and prevents sub-buffering. If pooling or lifetime
+ tracking is required, memories should be allocated through a custom
+ GstAllocator instead of relying on the lifetime of the buffers the
+ memories were originally attached to, which is fragile anyway.
+
+- The GstPlayer high-level playback library is being replaced with the
+ new GstPlay library (see above). GstPlayer should be considered
+ deprecated at this point and will be marked as such in the next
+ development cycle. Applications should be ported to GstPlay.
+
+- Gstreamer Editing Services: ges_video_transition_set_border(),
+ ges_video_transition_get_border()
+ ges_video_transition_set_inverted()
+ ges_video_transition_is_inverted() have been deprecated, use
+ ges_timeline_element_set_children_properties() instead.
Miscellaneous performance, latency and memory optimisations
-- this section will be filled in in due course
+More video conversion fast paths
-Miscellaneous other changes and enhancements
+- v210 ↔ I420, YV12, Y42B, UYVY and YUY2
+- A420 → RGB
+
+Less jitter when waiting on the system clock
+
+- Better system clock wait accuracy, less jitter: where available,
+ clock_nanosleep is used for higher accuracy for waits below 500
+ usecs, and waits below 2ms will first use the regular waiting system
+ and then clock_nanosleep for the remainder. The various wait
+ implementation have a latency ranging from 50 to 500+ microseconds.
+ While this is not a major issue when dealing with a low number of
+ waits per second (for ex: video), it does introduce a non-negligible
+ jitter for synchronisation of higher packet rate systems.
+
+Video decoder subframe support
-- this section will be filled in in due course
+- The GstVideoDecoder base class gained API to process input at the
+ sub-frame level. That way video decoders can start decoding slices
+ before they have received the full input frame in its entirety (to
+ the extent this is supported by the codec, of course). This helps
+ with CPU utilisation and reduces latency.
-Tracing framework and debugging improvements
+- This functionality is now being used in the OpenJPEG JPEG 2000
+ decoder, the FFmpeg H.264 decoder (in case of NAL-aligned input) and
+ the OpenMAX H.264/H.265 decoders (in case of NAL-aligned input).
-- this section will be filled in in due course
+Miscellaneous other changes and enhancements
+
+- GstDeviceMonitor no longer fails to start just because one of the
+ device providers failed to start. That could happen for example on
+ systems where the pulseaudio device provider is installed, but
+ pulseaudio isn’t actually running but ALSA is used for audio
+ instead. In the same vein the device monitor now keeps track of
+ which providers have been started (via the new
+ gst_device_provider_is_started()) and only stops actually running
+ device providers when stopping the device monitor.
+
+- On embedded systems it can be useful to create a registry that can
+ be shared and read by multiple processes running as different users.
+ It is now possible to set the new GST_REGISTRY_MODE environment
+ variable to specify the file mode for the registry file, which by
+ default is set to be only user readable/writable.
+
+- GstNetClientClock will signal lost sync in case the remote time
+ resets (e.g. because device power cycles), by emitting the “synced”
+ signal with synced=FALSE parameter, so applications can take action.
+
+- gst_value_deserialize_with_pspec() allows deserialization with a
+ hint for what the target GType should be. This allows for example
+ passing arrays of flags through the command line or
+ gst_util_set_object_arg(), eg: foo="<bar,bar+baz>".
+
+- It’s now allowed to create an empty GstVideoOverlayComposition
+ without any rectangles by passing a NULL rectangle to
+ gst_video_overlay_composition_new(). This is useful for bindings and
+ simplifies application code in some places.
+
+Tracing framework, debugging and testing improvements
+
+- New factories tracer to list loaded elements (and other plugin
+ features). This can be useful to collect a list of elements needed
+ for an application, which then in turn can be used to create a
+ tailored minimal GStreamer build that contains just the elements
+ needed and nothing else.
+- New plugin-feature-loaded tracing hook for use by tracers like the
+ new factories tracer
+
+- GstHarness: Add gst_harness_set_live() so that harnesses can be set
+ to non-live and return is-live=false in latency queries if needed.
+ Default behaviour is to always return is-live=true in latency
+ queries.
+
+- navseek: new "hold-eos" property. When enabled, the element will
+ hold back an EOS event until the next keystroke (via navigation
+ events). This can be used to keep a video sink showing the last
+ frame of a video pipeline until a key is pressed instead of tearing
+ it down immediately on EOS.
+
+- New fakeaudiosink element: mimics an audio sink and can be used for
+ testing and CI pipelines on systems where no audio system is
+ installed or running. It differs from fakesink in that it only
+ support audio caps and syncs to the clock by default like a normal
+ audio sink. It also implements the GstStreamVolume interface like
+ most audio sinks do.
+
+- New videocodectestsink element for video codec conformance testing:
+ Calculates MD5 checksums for video frames and skips any padding
+ whilst doing so. Can optionally also write back the video data with
+ padding removed into a file for easy byte-by-byte comparison with
+ reference data.
Tools
-- this section will be filled in in due course
+gst-inspect-1.0
+
+- Can sort the list of plugins by passing --sort=name as command line
+ option
+
+gst-launch-1.0
+
+- will now error out on top-level properties that don’t exist and
+ which were silently ignored before
+- On Windows the high-resolution clock is enabled now, which provides
+ better clock and timer performance on Windows (see Windows section
+ below for more details).
+
+gst-play-1.0
+
+- New --start-position command line argument to start playback from
+ the specified position
+- Audio can be muted/unmuted in interactive mode by pressing the m
+ key.
+- On Windows the high-resolution clock is enabled now (see Windows
+ section below for more details)
+
+gst-device-monitor-1.0
+
+- New --include-hidden command line argument to also show “hidden”
+ device providers
+
+ges-launch-1.0
+
+- New interactive mode that allows seeking and such. Can be disabled
+ by passing the --no-interactive argument on the command line.
+- Option to forward tags
+- Allow using an existing clip to determine the rendering format (both
+ topology and profile) via new --profile-from command line argument.
GStreamer RTSP server
-- this section will be filled in in due course
+- GstRTSPMediaFactory gained API to disable RTCP
+ (gst_rtsp_media_factory_set_enable_rtcp(), "enable-rtcp" property).
+ Previously RTCP was always allowed for all RTSP medias. With this
+ change it is possible to disable RTCP completely, no matter if the
+ client wants to do RTCP or not.
+
+- Make a mount point of / work correctly. While not allowed by the
+ RTSP 2 spec, the RTSP 1 spec is silent on this and it is used in the
+ wild. It is now possible to use / as a mount path in
+ gst-rtsp-server, e.g. rtsp://example.com/ would work with this now.
+ Note that query/fragment parts of the URI are not necessarily
+ correctly handled, and behaviour will differ between various
+ client/server implementations; so use it if you must but don’t bug
+ us if it doesn’t work with third party clients as you’d hoped.
+
+- multithreading fixes (races, refcounting issues, deadlocks)
+
+- ONVIF audio backchannel fixes
+
+- ONVIF trick mode optimisations
+
+- rtspclientsink: new "update-sdp" signal that allows updating the SDP
+ before sending it to the server via ANNOUNCE. This can be used to
+ add additional metadata to the SDP, for example. The order and
+ number of medias must not be changed, however.
GStreamer VAAPI
-- this section will be filled in in due course
+- new AV1 decoder element (vaapiav1dec)
+
+- H264 decoder: handle stereoscopic 3D video with frame packing
+ arrangement SEI messages
+
+- H265 encoder: added Screen Content Coding extensions support
+
+- H265 decoder: gained MAIN_444_12 profile support (decoded to
+ Y412_LE), and 4:2:2 12-bits support (decoded to Y212_LE)
+
+- vaapipostproc: gained BT2020 color standard support
+
+- vaapidecode: now generates caps templates dynamically at runtime in
+ order to advertise actually supported caps instead of all
+ theoretically supported caps.
+
+- GST_VAAPI_DRM_DEVICE environment variable to force a specified DRM
+ device when a DRM display is used. It is ignored when other types of
+ displays are used. By default /dev/dri/renderD128 is used for DRM
+ display.
GStreamer OMX
-- this section will be filled in in due course
+- subframe support in H.264/H.265 decoders
GStreamer Editing Services and NLE
-- this section will be filled in in due course
+- framepositioner: new "operator" property to access blending modes in
+ the compositor
+- timeline: Implement snapping to markers
+- smart-mixer: Add support for d3d11compositor and glvideomixer
+- titleclip: add "draw-shadow" child property
+- ges:// URI support to define a timeline from a description.
+- command-line-formatter
+ - Add track management to timeline description
+ - Add keyframe support
+- ges-launch-1.0:
+ - Add an interactive mode where we can seek etc…
+ - Add option to forward tags
+ - Allow using an existing clip to determine the rendering format
+ (both topology and profile) via new --profile-from command line
+ argument.
+- Fix static build
GStreamer validate
-- this section will be filled in in due course
+- report: Add a way to force backtraces on reports even if not a
+ critical issue (GST_VALIDATE_ISSUE_FLAGS_FORCE_BACKTRACE)
+- Add a flag to gst_validate_replace_variables_in_string() allow
+ defining how to resolve variables in structs
+- Add gst_validate_bin_monitor_get_scenario() to get the bin monitor
+ scenario, which is useful for applications that use Validate
+ directly.
+- Add an expected-values parameter to wait, message-type=XX allowing
+ more precise filtering of the message we are waiting for.
+- Add config file support: each test can now use a config file for the
+ given media file used to test.
+- Add support to check properties of object properties
+- scenario: Add an "action-done" signal to signal when an action is
+ done
+- scenario: Add a "run-command" action type
+- scenario: Allow forcing running action on idle from scenario file
+- scenario: Allow iterating over arrays in foreach
+- scenario: Rename ‘interlaced’ action to ‘non-blocking’
+- scenario: Add a non-blocking flag to the wait signal
GStreamer Python Bindings
-- this section will be filled in in due course
+- Fixes for Python 3.10
+- Various build fixes
+- at least one known breaking change caused by g-i annotation changes
+ (see below)
GStreamer C# Bindings
-- this section will be filled in in due course
+- Fix GstDebugGraphDetails enum
+- Updated to latests GtkSharp
+- Updated to include GStreamer 1.20 API
GStreamer Rust Bindings and Rust Plugins
-The GStreamer Rust bindings are released separately with a different
-release cadence that’s tied to gtk-rs, but the latest release has
-already been updated for the upcoming new GStreamer 1.20 API.
-
-gst-plugins-rs, the module containing GStreamer plugins written in Rust,
-has also seen lots of activity with many new elements and plugins.
-
-What follows is a list of elements and plugins available in
-gst-plugins-rs, so people don’t miss out on all those potentially useful
-elements that have no C equivalent.
-
-- FIXME: add new elements
-
-Rust audio plugins
-
-- audiornnoise: New element for audio denoising which implements the
- noise removal algorithm of the Xiph RNNoise library, in Rust
-- rsaudioecho: Port of the audioecho element from gst-plugins-good
- rsaudioloudnorm: Live audio loudness normalization element based on
- the FFmpeg af_loudnorm filter
-- claxondec: FLAC lossless audio codec decoder element based on the
- pure-Rust claxon implementation
-- csoundfilter: Audio filter that can use any filter defined via the
- Csound audio programming language
-- lewtondec: Vorbis audio decoder element based on the pure-Rust
- lewton implementation
-
-Rust video plugins
-
-- cdgdec/cdgparse: Decoder and parser for the CD+G video codec based
- on a pure-Rust CD+G implementation, used for example by karaoke CDs
-- cea608overlay: CEA-608 Closed Captions overlay element
-- cea608tott: CEA-608 Closed Captions to timed-text (e.g. VTT or SRT
- subtitles) converter
-- tttocea608: CEA-608 Closed Captions from timed-text converter
-- mccenc/mccparse: MacCaption Closed Caption format encoder and parser
-- sccenc/sccparse: Scenarist Closed Caption format encoder and parser
-- dav1dec: AV1 video decoder based on the dav1d decoder implementation
- by the VLC project
-- rav1enc: AV1 video encoder based on the fast and pure-Rust rav1e
- encoder implementation
-- rsflvdemux: Alternative to the flvdemux FLV demuxer element from
- gst-plugins-good, not feature-equivalent yet
-- rsgifenc/rspngenc: GIF/PNG encoder elements based on the pure-Rust
- implementations by the image-rs project
-
-Rust text plugins
-
-- textwrap: Element for line-wrapping timed text (e.g. subtitles) for
- better screen-fitting, including hyphenation support for some
- languages
-
-Rust network plugins
-
-- reqwesthttpsrc: HTTP(S) source element based on the Rust
- reqwest/hyper HTTP implementations and almost feature-equivalent
- with the main GStreamer HTTP source souphttpsrc
-- s3src/s3sink: Source/sink element for the Amazon S3 cloud storage
-- awstranscriber: Live audio to timed text transcription element using
- the Amazon AWS Transcribe API
-
-Generic Rust plugins
-
-- sodiumencrypter/sodiumdecrypter: Encryption/decryption element based
- on libsodium/NaCl
-- togglerecord: Recording element that allows to pause/resume
- recordings easily and considers keyframe boundaries
-- fallbackswitch/fallbacksrc: Elements for handling potentially
- failing (network) sources, restarting them on errors/timeout and
- showing a fallback stream instead
-- threadshare: Set of elements that provide alternatives for various
- existing GStreamer elements but allow to share the streaming threads
- between each other to reduce the number of threads
-- rsfilesrc/rsfilesink: File source/sink elements as replacements for
- the existing filesrc/filesink elements
+- The GStreamer Rust bindings are released separately with a different
+ release cadence that’s tied to gtk-rs, but the latest release has
+ already been updated for the upcoming new GStreamer 1.20 API (v1_20
+ feature).
+
+- gst-plugins-rs, the module containing GStreamer plugins written in
+ Rust, has also seen lots of activity with many new elements and
+ plugins. See the New Elements section above for a list of new Rust
+ elements.
Build and Dependencies
- Meson 0.59 or newer is required to build GStreamer now.
-- FIXME: this section will be filled in in due course
+- The GLib requirement has been bumped to GLib 2.56 or newer (from
+ March 2018).
+
+- The wpe plugin now requires wpe >= 2.28 and wpebackend-fdo >= 1.8
Explicit opt-in required for build of certain plugins with (A)GPL dependencies
See Building plugins with (A)GPL-licensed dependencies for more details
and a non-exhaustive list of plugins affected.
-gst-build: replaced by Monorepo
+gst-build: replaced by mono repository
-- this section will be filled in in due course
-
-- FIXME: describe + link to Monorepo FAQ
+See mono repository section above and the GStreamer mono repository FAQ.
Cerbero
on platforms where dependencies are not readily available, such as
Windows, Android, iOS and macOS.
-General improvements
+General Cerbero improvements
-- this section will be filled in in due course
+- Plugin removed: libvisual
+- New plugins: rtpmanagerbad and rist
-macOS / iOS
+macOS / iOS specific Cerbero improvements
-- this section will be filled in in due course
+- XCode 12 support
+- macOS OS release support is now future-proof, similar to iOS
+- macOS Apple Silicon (ARM64) cross-compile support has been added
+- macOS Apple Silicon (ARM64) native support is currently experimental
-Windows
+Windows specific Cerbero improvements
-- this section will be filled in in due course
+- Visual Studio 2022 support has been added
+- bootstrap is faster since it requires building fewer build-tools
+ recipes on Windows
+- package is faster due to better scheduling of recipe stages and
+ elimination of unnecessary autotools regeneration
+- The following plugins are no longer built on Windows:
+ - a52dec (another decoder is still available in libav)
+ - dvdread
+ - resindvd
Windows MSI installer
-- this section will be filled in in due course
+- no major changes
-Linux
+Linux specific Cerbero improvements
-- this section will be filled in in due course
+- Fedora, Debian OS release support is now more future-proof
+- Amazon Linux 2 support has been added
-Android
+Android specific Cerbero improvements
-- this section will be filled in in due course
+- no major changes
Platform-specific changes and improvements
Android
-- this section will be filled in in due course
+- No major changes
macOS and iOS
-- this section will be filled in in due course
+- applemedia: add ProRes support to vtenc and vtdec
Windows
-- this section will be filled in in due course
+- On Windows the high-resolution clock is enabled now in the
+ gst-launch-1.0 and gst-play-1.0 command line tools, which provides
+ better clock and timer performance on Windows, at the cost of higher
+ power consumption. By default, without the high-resolution clock
+ enabled, the timer precision on Windows is system-dependent and may
+ be as bad as 15ms which is not good enough for many multimedia
+ applications. Developers may want to do the same in their Windows
+ applications if they think it’s a good idea for their application
+ use case, and depending on the Windows version they target. This is
+ not done automatically by GStreamer because on older Windows
+ versions (pre-Windows 10) this affects a global Windows setting and
+ also there’s a power consumption vs. performance trade-off that may
+ differ from application to application.
+
+- dxgiscreencapsrc now supports resolution changes
+
+- The wasapi2 audio plugin was rewritten and now has a higher rank
+ than the old wasapi plugin since it has a number of additional
+ features such as automatic stream routing, and no
+ known-but-hard-to-fix issues. The plugin is always built if the
+ Windows 10 SDK is available now.
+
+- The wasapi device providers now detect and notify dynamic device
+ additions/removals
+
+- d3d11screencapturesrc: new desktop capture element, including
+ GstDeviceProvider implementation to enumerate/select target monitors
+ for capture.
+
+- Direct3D11/DXVA decoder now supports AV1 and MPEG2 codecs
+ (d3d11av1dec, d3d11mpeg2dec)
+
+- VP9 decoding got more reliable and stable thanks to a newly written
+ codec parser
+
+- Support for decoding interlaced H.264/AVC streams
+
+- Hardware-accelerated video deinterlacing (d3d11deinterlace) and
+ video mixing (d3d11compositor)
+
+- Video mixing with the Direct3D11 API (d3d11compositor)
+
+- MediaFoundation API based hardware encoders gained the ability to
+ receive Direct3D11 textures as an input
+
+- Seungha’s blog post “GStreamer ❤ Windows: A primer on the cool stuff
+ you’ll find in the 1.20 release” describes many of the
+ Windows-related improvements in more detail
Linux
-- this section will be filled in in due course
+- bluez: LDAC Bluetooth audio codec support in a2dpsink and avdtpsink,
+ as well as an LDAC RTP payloader (rtpldacpay) and an LDAC audio
+ encoder (ldacenc)
+
+- kmssink: gained support for NV24, NV61, RGB16/BGR16 formats;
+ auto-detect NVIDIA Tegra driver
Documentation improvements
-- this section will be filled in in due course
+- hardware-accelerated GPU plugins will now no longer always list all
+ the element variants for all available GPUs, since those are
+ system-dependent and it’s confusing for users to see those in the
+ documentation just because the GStreamer developer who generated the
+ docs had multiple GPUs to play with at the time. Instead just show
+ the default elements.
-Possibly Breaking Changes
+Possibly Breaking and Other Noteworthy Behavioural Changes
+
+- gst_parse_launch(), gst_parse_bin_from_description() and friends
+ will now error out when setting properties that don’t exist on
+ top-level bins. They were silently ignored before.
+
+- The GstWebRTC library does not expose any objects anymore with
+ public fields. Instead properties have been added to replace that
+ functionality. If you are accessing such fields in your application,
+ switch to the corresponding properties.
-- FIXME: this section will be filled in in due course
-- MPEG-TS SCTE-35 API changes (FIXME: flesh out)
-- gst_parse_launch() and friends now error out on non-existing
- properties on top-level bins where they would silently fail and
- ignore those before.
- playbin and uridecodebin now emit the source-setup signal before the
element is added to the bin and linked so that the source element is
already configured before any scheduling query comes in, which is
- useful for elements such as appsrc or giostreamsrc. (Merge Request)
+ useful for elements such as appsrc or giostreamsrc.
+
+- The source element inside urisourcebin (used inside uridecodebin3
+ which is used inside playbin3) is no longer called "source". This
+ shouldn’t affect anyone hopefully, because there’s a "setup-source"
+ signal to configure the source element and no one should rely on
+ names of internal elements anyway.
+
+- The vp8enc element now expects bps (bits per second) for the
+ "temporal-scalability-target-bitrate" property, which is consistent
+ with the "target-bitrate" property. Since additional configuration
+ is required with modern libvpx to make temporal scaling work anyway,
+ chances are that very few people will have been using this property
+
+- vp8enc and vp9enc now default to “good quality” for the "deadline"
+ property rather then “best quality”. Having the deadline set to best
+ quality causes the encoder to be absurdly slow, most real-life users
+ will want the good quality tradeoff instead.
+
+- The experimental GstTranscoder library API in gst-plugins-bad was
+ changed from a GObject signal-based notification mechanism to a
+ GstBus/message-based mechanism akin to GstPlayer/GstPlay.
+
+- MPEG-TS SCTE-35 API: semantic change for SCTE-35 splice commands:
+ timestamps passed by the application should be in running time now,
+ since users of the API can’t really be expected to predict the local
+ PTS of the muxer.
+
+- The GstContext used by souphttpsrc to share the session between
+ multiple element instances has changed. Previously it provided
+ direct access to the internal SoupSession object, now it only
+ provides access to an opaque, internal type. This change is
+ necessary because SoupSession is not thread-safe at all and can’t be
+ shared safely between arbitrary external code and souphttpsrc.
+
+- Python bindings: GObject-introspection related Annotation fixes have
+ led to a case of a GstVideo.VideoInfo-related function signature
+ changing in the Python bindings (possibly one or two other cases
+ too). This is for a function that should never have been exposed in
+ the first place though, so the bindings are being updated to throw
+ an exception in that case, and the correct replacement API has been
+ added in form of an override.
Known Issues
-- this section will be filled in in due course
-
-- There are a couple of known WebRTC-related regressions/blockers:
-
- - webrtc: DTLS setup with Chrome is broken
- - webrtcbin: First keyframe is usually lost
+- nothing in particular at this point (but also see possibly breaking
+ changes section above)
Contributors
-- this section will be filled in in due course
+Aaron Boxer, Adam Leppky, Adam Williamson, Alba Mendez, Alejandro
+González, Aleksandr Slobodeniuk, Alexander Vandenbulcke, Alex Ashley,
+Alicia Boya García, Andika Triwidada, Andoni Morales Alastruey, Andrew
+Wesie, Andrey Moiseev, Antonio Ospite, Antonio Rojas, Arthur Crippa
+Búrigo, Arun Raghavan, Ashley Brighthope, Axel Kellermann, Baek, Bastien
+Nocera, Bastien Reboulet, Benjamin Gaignard, Bing Song, Binh Truong,
+Biswapriyo Nath, Brad Hards, Brad Smith, Brady J. Garvin, Branko
+Subasic, Camilo Celis Guzman, Chris Bass, ChrisDuncanAnyvision, Chris
+White, Corentin Damman, Daniel Almeida, Daniel Knobe, Daniel Stone,
+david, David Fernandez, David Keijser, David Phung, Devarsh Thakkar,
+Dinesh Manajipet, Dmitry Samoylov, Dmitry Shusharin, Dominique Martinet,
+Doug Nazar, Ederson de Souza, Edward Hervey, Emmanuel Gil Peyrot,
+Enrique Ocaña González, Ezequiel Garcia, Fabian Orccon, Fabrice
+Fontaine, Fernando Jimenez Moreno, Florian Karydes, Francisco Javier
+Velázquez-García, François Laignel, Frederich Munch, Fredrik Pålsson,
+George Kiagiadakis, Georg Lippitsch, Göran Jönsson, Guido Günther,
+Guillaume Desmottes, Guiqin Zou, Haakon Sporsheim, Haelwenn (lanodan)
+Monnier, Haihao Xiang, Haihua Hu, Havard Graff, He Junyan, Helmut
+Januschka, Henry Wilkes, Hosang Lee, Hou Qi, Ignacio Casal Quinteiro,
+Igor Kovalenko, Ilya Kreymer, Imanol Fernandez, Jacek Tomaszewski, Jade
+Macho, Jakub Adam, Jakub Janků, Jan Alexander Steffens (heftig), Jan
+Schmidt, Jason Carrete, Jason Pereira, Jay Douglass, Jeongki Kim, Jérôme
+Laheurte, Jimmi Holst Christensen, Johan Sternerup, John Hassell, John
+Lindgren, John-Mark Bell, Jonathan Matthew, Jordan Petridis, Jose
+Quaresma, Julian Bouzas, Julien, Kai Uwe Broulik, Kasper Steensig
+Jensen, Kellermann Axel, Kevin Song, Khem Raj, Knut Inge Hvidsten, Knut
+Saastad, Kristofer Björkström, Lars Lundqvist, Lawrence Troup, Lim Siew
+Hoon, Lucas Stach, Ludvig Rappe, Luis Paulo Fernandes de Barros, Luke
+Yelavich, Mads Buvik Sandvei, Marc Leeman, Marco Felsch, Marek Vasut,
+Marian Cichy, Marijn Suijten, Marius Vlad, Markus Ebner, Mart Raudsepp,
+Matej Knopp, Mathieu Duponchelle, Matthew Waters, Matthieu De Beule,
+Mengkejiergeli Ba, Michael de Gans, Michael Olbrich, Michael Tretter,
+Michal Dzik, Miguel Paris, Mikhail Fludkov, mkba, Nazar Mokrynskyi,
+Nicholas Jackson, Nicola Murino, Nicolas Dufresne, Niklas Hambüchen,
+Nikolay Sivov, Nirbheek Chauhan, Olivier Blin, Olivier Crete, Olivier
+Crête, Paul Goulpié, Per Förlin, Peter Boba, P H, Philippe Normand,
+Philipp Zabel, Pieter Willem Jordaan, Piotrek Brzeziński, Rafał
+Dzięgiel, Rafostar, raghavendra, Raghavendra, Raju Babannavar, Raleigh
+Littles III, Randy Li, Randy Li (ayaka), Ratchanan Srirattanamet, Raul
+Tambre, reed.lawrence, Ricky Tang, Robert Rosengren, Robert Swain, Robin
+Burchell, Roman Sivriver, R S Nikhil Krishna, Ruben Gonzalez, Ruslan
+Khamidullin, Sanchayan Maity, Scott Moreau, Sebastian Dröge, Sergei
+Kovalev, Seungha Yang, Sid Sethupathi, sohwan.park, Sonny Piers, Staz M,
+Stefan Brüns, Stéphane Cerveau, Stephan Hesse, Stian Selnes, Stirling
+Westrup, Théo MAILLART, Thibault Saunier, Tim, Timo Wischer, Tim-Philipp
+Müller, Tim Schneider, Tobias Ronge, Tom Schoonjans, Tulio Beloqui,
+tyler-aicradle, U. Artie Eoff, Ung, Val Doroshchuk, VaL Doroshchuk,
+Víctor Manuel Jáquez Leal, Vivek R, Vivia Nikolaidou, Vivienne
+Watermeier, Vladimir Menshakov, Will Miller, Wim Taymans, Xabier
+Rodriguez Calvar, Xavier Claessens, Xℹ Ruoyao, Yacine Bandou, Yinhang
+Liu, youngh.lee, youngsoo.lee, yychao, Zebediah Figura, Zhang yuankun,
+Zhang Yuankun, Zhao, Zhao Zhili, , Aleksandar Topic, Antonio Ospite,
+Bastien Nocera, Benjamin Gaignard, Brad Hards, Carlos Falgueras García,
+Célestin Marot, Corentin Damman, Corentin Noël, Daniel Almeida, Daniel
+Knobe, Danny Smith, Dave Piché, Dmitry Osipenko, Fabrice Fontaine,
+fjmax, Florian Zwoch, Guillaume Desmottes, Haihua Hu, Heinrich Kruger,
+He Junyan, Jakub Adam, James Cowgill, Jan Alexander Steffens (heftig),
+Jean Felder, Jeongki Kim, Jiri Uncovsky, Joe Todd, Jordan Petridis,
+Krystian Wojtas, Marc-André Lureau, Marcin Kolny, Marc Leeman, Mark
+Nauwelaerts, Martin Reboredo, Mathieu Duponchelle, Matthew Waters,
+Mengkejiergeli Ba, Michael Gruner, Nicolas Dufresne, Nirbheek Chauhan,
+Olivier Crête, Philippe Normand, Rafał Dzięgiel, Ralf Sippl, Robert
+Mader, Sanchayan Maity, Sangchul Lee, Sebastian Dröge, Seungha Yang,
+Stéphane Cerveau, Teh Yule Kim, Thibault Saunier, Thomas Klausner, Timo
+Wischer, Tim-Philipp Müller, Tobias Reineke, Tomasz Andrzejak, Trung Do,
+Tyler Compton, Ung, Víctor Manuel Jáquez Leal, Vivia Nikolaidou, Wim
+Taymans, wngecn, Wonchul Lee, wuchang li, Xavier Claessens, Xi Ruoyao,
+Yoshiharu Hirose, Zhao,
… and many others who have contributed bug reports, translations, sent
suggestions or helped testing.
1.20.0
-1.20.0 is scheduled to be released around October/November 2021.
+1.20.0 is scheduled to be released around early February 2022.
Schedule for 1.22
unstable development version leading up to the stable 1.22 release. The
development of 1.21/1.22 will happen in the git main branch.
-The plan for the 1.22 development cycle is yet to be confirmed.
+The plan for the 1.22 development cycle is yet to be confirmed. Assuming
+no major project-wide reorganisations in the 1.22 cycle we might try and
+aim for a release around August 2022.
1.22 will be backwards-compatible to the stable 1.20, 1.18, 1.16, 1.14,
1.12, 1.10, 1.8, 1.6, 1.4, 1.2 and 1.0 release series.
------------------------------------------------------------------------
These release notes have been prepared by Tim-Philipp Müller with
-contributions from …
+contributions from Matthew Waters, Nicolas Dufresne, Nirbheek Chauhan,
+Sebastian Dröge and Seungha Yang.
License: CC BY-SA 4.0
-This is GStreamer gst-plugins-good 1.19.3.
+This is GStreamer gst-plugins-good 1.19.90.
GStreamer 1.19 is the development branch leading up to the next major
stable version which will be 1.20.
"construct": false,
"construct-only": false,
"controllable": false,
- "default": "GStreamer 1.19.3.1 FLV muxer",
+ "default": "GStreamer 1.19.90 FLV muxer",
"mutable": "null",
"readable": true,
"type": "gchararray",
"construct": false,
"construct-only": false,
"controllable": false,
- "default": "GStreamer 1.19.3.1 FLV muxer",
+ "default": "GStreamer 1.19.90 FLV muxer",
"mutable": "null",
"readable": true,
"type": "gchararray",
"construct": false,
"construct-only": false,
"controllable": false,
- "default": "GStreamer/1.19.3.1",
+ "default": "GStreamer/1.19.90",
"mutable": "null",
"readable": true,
"type": "gchararray",
"construct": false,
"construct-only": false,
"controllable": false,
- "default": "GStreamer souphttpsrc 1.19.3.1 ",
+ "default": "GStreamer souphttpsrc 1.19.90 ",
"mutable": "null",
"readable": true,
"type": "gchararray",
<release>
<Version>
+ <revision>1.19.90</revision>
+ <branch>main</branch>
+ <name></name>
+ <created>2022-01-28</created>
+ <file-release rdf:resource="https://gstreamer.freedesktop.org/src/gst-plugins-good/gst-plugins-good-1.19.90.tar.xz" />
+ </Version>
+ </release>
+
+ <release>
+ <Version>
<revision>1.19.3</revision>
<branch>main</branch>
<name></name>
project('gst-plugins-good', 'c',
- version : '1.19.3.1',
+ version : '1.19.90',
meson_version : '>= 0.59',
default_options : [ 'warning_level=1',
'buildtype=debugoptimized' ])
+=== release 1.19.90 ===
+
+2022-01-28 14:28:35 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * NEWS:
+ * RELEASE:
+ * gst-plugins-ugly.doap:
+ * meson.build:
+ Release 1.19.90
+
+2022-01-28 14:28:28 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * ChangeLog:
+ Update ChangeLogs for 1.19.90
+
2022-01-27 01:39:28 +0000 Tim-Philipp Müller <tim@centricular.com>
* po/LINGUAS:
GStreamer 1.20 Release Notes
GStreamer 1.20 has not been released yet. It is scheduled for release in
-November 2021.
+late January / early February 2022.
1.19.x is the unstable development version that is being developed in
-the git main branch and which will eventually result in 1.20, and 1.19.3
-is the current development release in that series
-
-It is expected that feature freeze will be in early October 2021,
-followed by one or two 1.19.9x pre-releases and the new 1.20 stable
-release around October/November 2021.
+the git main branch and which will eventually result in 1.20, and
+1.19.90 is the first release candidate in that series (1.20rc1).
1.20 will be backwards-compatible to the stable 1.18, 1.16, 1.14, 1.12,
-1.10, 1.8, 1.6,, 1.4, 1.2 and 1.0 release series.
+1.10, 1.8, 1.6, 1.4, 1.2 and 1.0 release series.
See https://gstreamer.freedesktop.org/releases/1.20/ for the latest
version of this document.
-Last updated: Monday 1 November 2021, 01:00 UTC (log)
+Last updated: Wednesday 26 January 2022, 01:00 UTC (log)
Introduction
Highlights
-- this section will be completed in due course
+- Development in GitLab was switched to a single git repository
+ containing all the modules
+- GstPlay: new high-level playback library, replaces GstPlayer
+- WebM Alpha decoding support
+- Encoding profiles can now be tweaked with additional
+ application-specified element properties
+- Compositor: multi-threaded video conversion and mixing
+- RTP header extensions: unified support in RTP depayloader and
+ payloader base classes
+- SMPTE 2022-1 2-D Forward Error Correction support
+- Smart encoding (passthrough) support for VP8, VP9, H.265 in
+ encodebin and transcodebin
+- Runtime compatibility support for libsoup2 and libsoup3 (libsoup3
+ support experimental)
+- Video decoder subframe support
+- Video decoder automatic packet-loss, data corruption, and keyframe
+ request handling for RTP / WebRTC / RTSP
+- MP4 and Matroska muxers now support profile/level/resolution changes
+ for H264/H265 input streams (i.e. codec data changing on the fly)
+- MP4 muxing mode that initially creates a fragmented mp4 which is
+ converted to a regular mp4 on EOS
+- Audio support for the WebKit Port for Embedded (WPE) web page source
+ element
+- CUDA based video color space convert and rescale elements and
+ upload/download elements
+- NVIDIA memory:NVMM support for OpenGL glupload and gldownload
+ elements
+- Many WebRTC improvements
+- The new VA-API plugin implemention fleshed out with more decoders
+ and new postproc elements
+- AppSink API to retrieve events in addition to buffers and buffer
+ lists
+- AppSrc gained more configuration options for the internal queue
+ (leakiness, limits in buffers and time, getters to read current
+ levels)
+- Updated Rust bindings and many new Rust plugins
+- Improved support for custom minimal GStreamer builds
+- Support build against FFmpeg 5.0
+- Linux Stateless CODEC support gained MPEG2 and VP9
+- Windows Direct3D11/DXVA decoder gained AV1 and MPEG2 support
+- Lots of new plugins, features, performance improvements and bug
+ fixes
Major new features and changes
Noteworthy new features and API
-- this section will be filled in in due course
+- gst_element_get_request_pad() has been deprecated in favour of the
+ newly-added gst_element_request_pad_simple() which does the exact
+ same thing but has a less confusing name that hopefully makes clear
+ that the function request a new pad rather than just retrieves an
+ already-existing request pad.
+
+Development in GitLab was switched to a single git repository containing all the modules
+
+The GStreamer multimedia framework is a set of libraries and plugins
+split into a number of distinct modules which are released independently
+and which have so far been developed in separate git repositories in
+freedesktop.org GitLab.
+
+In addition to these separate git repositories there was a gst-build
+module that would use the Meson build systems’s subproject feature to
+download each individual module and then build everything in one go. It
+would also provide an uninstalled development environment that made it
+easy to work on GStreamer and use or test versions other than the
+system-installed GStreamer version.
+
+All of these modules have now (as of 28 September 2021) been merged into
+a single git repository (“Mono repository” or “monorepo”) which should
+simplify development workflows and continuous integration, especially
+where changes need to be made to multiple modules at once.
+
+This mono repository merge will primarily affect GStreamer developers
+and contributors and anyone who has workflows based on the GStreamer git
+repositories.
+
+The Rust bindings and Rust plugins modules have not been merged into the
+mono repository at this time because they follow a different release
+cycle.
+
+The mono repository lives in the existing GStreamer core git repository
+in GitLab in the new main branch and all future development will happen
+on this branch.
+
+Modules will continue to be released as separate tarballs.
+
+For more details, please see the GStreamer mono repository FAQ.
+
+GstPlay: new high-level playback library replacing GstPlayer
+
+- GstPlay is a new high-level playback library that replaces the older
+ GstPlayer API. It is basically the same API as GstPlayer but
+ refactored to use bus messages for application notifications instead
+ of GObject signals. There is still a signal adapter object for those
+ who prefer signals. Since the existing GstPlayer API is already in
+ use in various applications, it didn’t seem like a good idea to
+ break it entirely. Instead a new API was added, and it is expected
+ that this new GstPlay API will be moved to gst-plugins-base in
+ future.
+
+- The existing GstPlayer API is scheduled for deprecation and will be
+ removed at some point in the future (e.g. in GStreamer 1.24), so
+ application developers are urged to migrate to the new GstPlay API
+ at their earliest convenience.
+
+WebM alpha decoding
+
+- Implement WebM alpha decoding (VP8/VP9 with alpha), which required
+ support and additions in various places. This is supported both with
+ software decoders and hardware-accelerated decoders.
+
+- VP8/VP9 don’t support alpha components natively in the codec, so the
+ way this is implemented in WebM is by encoding the alpha plane with
+ transparency data as a separate VP8/VP9 stream. Inside the WebM
+ container (a variant of Matroska) this is coded as a single video
+ track with the “normal” VP8/VP9 video data making up the main video
+ data and each frame of video having an encoded alpha frame attached
+ to it as extra data ("BlockAdditional").
+
+- matroskademux has been extended extract this per-frame alpha side
+ data and attach it in form of a GstVideoCodecAlphaMeta to the
+ regular video buffers. Note that this new meta is specific to this
+ VP8/VP9 alpha support and can’t be used to just add alpha support to
+ other codecs that don’t support it. Lastly, matroskademux also
+ advertises the fact that the streams contain alpha in the caps.
+
+- The new codecalpha plugin contains various bits of infrastructure to
+ support autoplugging and debugging:
+
+ - codecalphademux splits out the alpha stream from the metas on
+ the regular VP8/VP9 buffers
+ - alphacombine takes two decoded raw video streams (one alpha, one
+ the regular video) and combines it into a video stream with
+ alpha
+ - vp8alphadecodebin + vp9alphadecodebin are wrapper bins that use
+ the regular vp8dec and vp9dec software decoders to decode
+ regular and alpha streams and combine them again. To decodebin
+ these look like regular decoders which ju
+ - The V4L2 CODEC plugin has stateless VP8/VP9 decoders that can
+ decode both alpha and non-alpha stream with a single decoder
+ instance
+
+- A new AV12 video format was added which is basically NV12 with an
+ alpha plane, which is more convenient for many hardware-accelerated
+ decoders.
+
+- Watch Nicolas Dufresne’s LCA 2022 talk “Bringing WebM Alpha support
+ to GStreamer” for all the details and a demo.
+
+RTP Header Extensions Base Class and Automatic Header Extension Handling in RTP Payloaders and Depayloaders
+
+- RTP Header Extensions are specified in RFC 5285 and provide a way to
+ add small pieces of data to RTP packets in between the RTP header
+ and the RTP payload. This is often used for per-frame metadata,
+ extended timestamps or other application-specific extra data. There
+ are several commonly-used extensions specified in various RFCs, but
+ senders are free to put any kind of data in there, as long as sender
+ and receiver both know what that data is. Receivers that don’t know
+ about the header extensions will just skip the extra data without
+ ever looking at it. These header extensions can often be combined
+ with any kind of payload format, so may need to be supported by many
+ RTP payloader and depayloader elements.
+
+- Inserting and extracting RTP header extension data has so far been a
+ bit inconvenient in GStreamer: There are functions to add and
+ retrieve RTP header extension data from RTP packets, but nothing
+ works automatically, even for common extensions. People would have
+ to do the insertion/extraction either in custom elements
+ before/after the RTP payloader/depayloader, or inside pad probes,
+ which isn’t very nice.
+
+- This release adds various pieces of new infrastructure for generic
+ RTP header extension handling, as well as some implementations for
+ common extensions:
+
+ - GstRTPHeaderExtension is a new helper base class for reading and
+ writing RTP header extensions. Nominally this subclasses
+ GstElement, but only so these extensions are stored in the
+ registry where they can be looked up by URI or name. They don’t
+ have pads and don’t get added to the pipeline graph as an
+ element.
+
+ - "add-extension" and "clear-extension" action signals on RTP
+ payloaders and depayloaders for manual extension management
+
+ - The "request-extension" signal will be emitted if an extension
+ is encountered that requires explicit mapping by the application
+
+ - new "auto-header-extension" property on RTP payloaders and
+ depayloaders for automatic handling of known header extensions.
+ This is enabled by default. The extensions must be signalled via
+ caps / SDP.
+
+ - RTP header extension implementations:
+
+ - rtphdrextclientaudiolevel: Client-to-Mixer Audio Level
+ Indication (RFC 6464) (also see below)
+ - rtphdrextcolorspace: Color Space extension, extends RTP
+ packets with color space and high dynamic range (HDR)
+ information
+ - rtphdrexttwcc: Transport Wide Congestion Control support
+
+- gst_rtp_buffer_remove_extension_data() is a new helper function to
+ remove an RTP header extension from an RTP buffer
+
+- The existing gst_rtp_buffer_set_extension_data() now also supports
+ shrinking the extension data in size
+
+AppSink and AppSrc improvements
+
+- appsink: new API to pull events out of appsink in addition to
+ buffers and buffer lists.
+
+ There was previously no way for users to receive incoming events
+ from appsink properly serialised with the data flow, even if they
+ are serialised events. The reason for that is that the only way to
+ intercept events was via a pad probe on the appsink sink pad, but
+ there is also internal queuing inside of appsink, so it’s difficult
+ to ascertain the right order of everything in all cases.
+
+ There is now a new "new-serialized-event" signal which will be
+ emitted when there’s a new event pending (just like the existing
+ "new-sample" signal). The "emit-signals" property must be set to
+ TRUE in order to activate this (but it’s also fine to just pull from
+ the application thread without using the signals).
+
+ gst_app_sink_pull_object() and gst_app_sink_try_pull_object() can be
+ used to pull out either an event or a new sample carrying a buffer
+ or buffer list, whatever is next in the queue.
+
+ EOS events will be filtered and will not be returned. EOS handling
+ can be done the ususal way, same as with _pull_sample().
+
+- appsrc: allow configuration of internal queue limits in time and
+ buffers and add leaky mode.
+
+ There is internal queuing inside appsrc so the application thread
+ can push data into the element which will then be picked up by the
+ source element’s streaming thread and pushed into the pipeline from
+ that streaming thread. This queue is unlimited by default and until
+ now it was only possible to set a maximum size limit in bytes. When
+ that byte limit is reached, the pushing thread (application thread)
+ would be blocked until more space becomes available.
+
+ A limit in bytes is not particularly useful for many use cases, so
+ now it is possible to also configure limits in time and buffers
+ using the new "max-time" and "max-buffers" properties. Of course
+ there are also matching new read-only"current-level-buffers" and
+ "current-level-time properties" properties to query the current fill
+ level of the internal queue in time and buffers.
+
+ And as if that wasn’t enough the internal queue can also be
+ configured as leaky using the new "leaky-type" property. That way
+ when the queue is full the application thread won’t be blocked when
+ it tries to push in more data, but instead either the new buffer
+ will be dropped or the oldest data in the queue will be dropped.
+
+Better string serialization of nested GstCaps and GstStructures
+
+- New string serialisation format for structs and caps that can handle
+ nested structs and caps properly by using brackets to delimit nested
+ items (e.g. some-struct, some-field=[nested-struct, nested=true]).
+ Unlike the default format the new variant can also support more than
+ one level of nesting. For backwards-compatibility reasons the old
+ format is still output by default when serialising caps and structs
+ using the existing API. The new functions gst_caps_serialize() and
+ gst_structure_serialize() can be used to output strings in the new
+ format.
+
+Convenience API for custom GstMetas
+
+- New convenience API to register and create custom GstMetas:
+ gst_meta_register_custom() and gst_buffer_add_custom_meta(). Such
+ custom meta is backed by a GstStructure and does not require that
+ users of the API expose their GstMeta implementation as public API
+ for other components to make use of it. In addition, it provides a
+ simpler interface by ignoring the impl vs. api distinction that the
+ regular API exposes. This new API is meant to be the meta
+ counterpart to custom events and messages, and to be more convenient
+ than the lower-level API when the absolute best performance isn’t a
+ requirement. The reason it’s less performant than a “proper” meta is
+ that a proper meta is just a C struct in the end whereas this goes
+ through the GstStructure API which has a bit more overhead, which
+ for most scenarios is negligible however. This new API is useful for
+ experimentation or proprietary metas, but also has some limitations:
+ it can only be used if there’s a single producer of these metas;
+ it’s not allowed to register the same custom meta multiple times or
+ from multiple places.
+
+Additional Element Properties on Encoding Profiles
+
+- GstEncodingProfile: The new "element-properties" and
+ gst_encoding_profile_set_element_properties() API allows
+ applications to set additional element properties on encoding
+ profiles to configure muxers and encoders. So far the encoding
+ profile template was the only place where this could be specified,
+ but often what applications want to do is take a ready-made encoding
+ profile shipped by GStreamer or the application and then tweak the
+ settings on top of that, which is now possible with this API. Since
+ applications can’t always know in advance what encoder element will
+ be used in the end, it’s even possible to specify properties on a
+ per-element basis.
+
+ Encoding Profiles are used in the encodebin, transcodebin and
+ camerabin elements and APIs to configure output formats (containers
+ and elementary streams).
+
+Audio Level Indication Meta for RFC 6464
+
+- New GstAudioLevelMeta containing Audio Level Indication as per RFC
+ 6464
+
+- The level element has been updated to add GstAudioLevelMeta on
+ buffers if the "audio-level-meta" property is set to TRUE. This can
+ then in turn be picked up by RTP payloaders to signal the audio
+ level to receivers through RTP header extensions (see above).
+
+- New Client-to-Mixer Audio Level Indication (RFC6464) RTP Header
+ Extension which should be automatically created and used by RTP
+ payloaders and depayloaders if their "auto-header-extension"
+ property is enabled and if the extension is part of the RTP caps.
+
+Automatic packet loss, data corruption and keyframe request handling for video decoders
+
+- The GstVideoDecoder base class has gained various new APIs to
+ automatically handle packet loss and data corruption better by
+ default, especially in RTP, RTSP and WebRTC streaming scenarios, and
+ to give subclasses more control about how they want to handle
+ missing data:
+
+ - Video decoder subclasses can mark output frames as corrupted via
+ the new GST_VIDEO_CODEC_FRAME_FLAG_CORRUPTED flag
+
+ - A new "discard-corrupted-frames" property allows applications to
+ configure decoders so that corrupted frames are directly
+ discarded instead of being forwarded inside the pipeline. This
+ is a replacement for the "output-corrupt" property of the FFmpeg
+ decoders.
+
+ - RTP depayloaders can now signal to decoders that data is missing
+ when sending GAP events for lost packets. GAP events can be sent
+ for various reason in a GStreamer pipeline. Often they are just
+ used to let downstream elements know that there isn’t a buffer
+ available at the moment, so downstream elements can move on
+ instead of waiting for one. They are also sent by RTP
+ depayloaders in the case that packets are missing, however, and
+ so far a decoder was not able to differentiate the two cases.
+ This has been remedied now: GAP events can be decorated with
+ gst_event_set_gap_flags() and GST_GAP_FLAG_MISSING_DATA to let
+ decoders now what happened, and decoders can then use that in
+ some cases to handle missing data better.
+
+ - The GstVideoDecoder::handle_missing_data vfunc was added to
+ inform subclasses about packet loss or missing data and let them
+ handle it in their own way if they like.
+
+ - gst_video_decoder_set_needs_sync_point() lets subclasses signal
+ that they need the stream to start with a sync point. If
+ enabled, the base class will discard all non-sync point frames
+ in the beginning and after a flush and does not pass them to the
+ subclass. Furthermore, if the first frame is not a sync point,
+ the base class will try and request a sync frame from upstream
+ by sending a force-key-unit event (see next items).
+
+ - New "automatic-request-sync-points" and
+ "automatic-request-sync-point-flags" properties to automatically
+ request sync points when needed, e.g. on packet loss or if the
+ first frame is not a keyframe. Applications may want to enable
+ this on decoders operating in e.g. RTP/WebRTC/RTSP receiver
+ pipelines.
+
+ - The new "min-force-key-unit-interval" property can be used to
+ ensure there’s a minimal interval between keyframe requests to
+ upstream (and/or the sender) and we’re not flooding the sender
+ with key unit requests.
+
+ - gst_video_decoder_request_sync_point() allows subclasses to
+ request a new sync point (e.g. if they choose to do their own
+ missing data handling). This will still honour the
+ "min-force-key-unit-interval" property if set.
+
+Improved support for custom minimal GStreamer builds
+
+- Element registration and registration of other plugin features
+ inside plugin init functions has been improved in order to
+ facilitate minimal custom GStreamer builds.
+
+- A number of new macros have been added to declare and create
+ per-element and per-pluginfeature register functions in all plugins,
+ and then call those from the per-plugin plugin_init functions:
+
+ - GST_ELEMENT_REGISTER_DEFINE,
+ GST_DEVICE_PROVIDER_REGISTER_DEFINE,
+ GST_DYNAMIC_TYPE_REGISTER_DEFINE, GST_TYPE_FIND_REGISTER_DEFINE
+ for the actual registration call with GStreamer
+ - GST_ELEMENT_REGISTER, GST_DEVICE_PROVIDER_REGISTER,
+ GST_DYNAMIC_TYPE_REGISTER, GST_PLUGIN_STATIC_REGISTER,
+ GST_TYPE_FIND_REGISTER to call the registration function defined
+ by the REGISTER_DEFINE macro
+ - GST_ELEMENT_REGISTER_DECLARE,
+ GST_DEVICE_PROVIDER_REGISTER_DECLARE,
+ GST_DYNAMIC_TYPE_REGISTER_DECLARE,
+ GST_TYPE_FIND_REGISTER_DECLARE to declare the registration
+ function defined by the REGISTER_DEFINE macro
+ - and various variants for advanced use cases.
+
+- This means that applications can call the per-element and
+ per-pluginfeature registration functions for only the elements they
+ need instead of registering plugins as a whole with all kinds of
+ elements that may not be required (e.g. encoder and decoder instead
+ of just decoder). In case of static linking all unused functions and
+ their dependencies would be removed in this case by the linker,
+ which helps minimise binary size for custom builds.
+
+- gst_init() will automatically call a gst_init_static_plugins()
+ function if one exists.
+
+- See the GStreamer static build documentation and Stéphane’s blog
+ post Generate a minimal GStreamer build, tailored to your needs for
+ more details.
New elements
-- this section will be filled in in due course
+- New aesdec and aesenc elements for AES encryption and decryption in
+ a custom format.
+
+- New encodebin2 element with dynamic/sometimes source pads in order
+ to support the option of doing the muxing outside of encodebin,
+ e.g. in combination with a splitmuxsink.
+
+- New fakeaudiosink and videocodectestsink elements for testing and
+ debugging (see below for more details)
+
+- rtpisacpay, rtpisacdepay: new RTP payloader and depayloader for iSAC
+ audio codec
+
+- rtpst2022-1-fecdec, rtpst2022-1-fecenc: new elements providing SMPTE
+ 2022-1 2-D Forward Error Correction. More details in Mathieu’s blog
+ post.
+
+- isac: new plugin wrapping the Internet Speech Audio Codec reference
+ encoder and decoder from the WebRTC project.
+
+- asio: plugin for Steinberg ASIO (Audio Streaming Input/Output) API
+
+- gssrc, gssink: add source and sink for Google Cloud Storage
+
+- onnx: new plugin to apply ONNX neural network models to video
+
+- openaptx: aptX and aptX-HD codecs using libopenaptx (v0.2.0)
+
+- qroverlay, debugqroverlay: new elements that allows overlaying data
+ on top of video in form of a QR code
+
+- cvtracker: new OpenCV-based tracker element
+
+- av1parse, vp9parse: new parsers for AV1 and VP9 video
+
+- va: work on the new VA-API plugin implementation for
+ hardware-accelerated video decoding and encoding has continued at
+ pace, with various new decoders and filters having joined the
+ initial vah264dec:
+
+ - vah265dec: VA-API H.265 decoder
+ - vavp8dec: VA-API VP8 decoder
+ - vavp9dec: VA-API VP9 decoder
+ - vaav1dec: VA-API AV1 decoder
+ - vampeg2dec: VA-API MPEG-2 decoder
+ - vadeinterlace: : VA-API deinterlace filter
+ - vapostproc: : VA-API postproc filter (color conversion,
+ resizing, cropping, color balance, video rotation, skin tone
+ enhancement, denoise, sharpen)
+
+ See Víctor’s blog post “GstVA in GStreamer 1.20” for more details
+ and what’s coming up next.
+
+- vaapiav1dec: new AV1 decoder element (in gstreamer-vaapi)
+
+- msdkav1dec: hardware-accelerated AV1 decoder using the Intel Media
+ SDK / oneVPL
+
+- nvcodec plugin for NVIDIA NVCODEC API for hardware-accelerated video
+ encoding and decoding:
+
+ - cudaconvert, cudascale: new CUDA based video color space convert
+ and rescale elements
+ - cudaupload, cudadownload: new helper elements for memory
+ transfer between CUDA and system memory spaces
+ - nvvp8sldec, nvvp9sldec: new GstCodecs-based VP8/VP9 decoders
+
+- Various new hardware-accelerated elements for Windows:
+
+ - d3d11screencapturesrc: new desktop capture element, including a
+ GstDeviceProvider implementation to enumerate/select target
+ monitors for capture.
+ - d3d11av1dec and d3d11mpeg2dec: AV1 and MPEG-2 decoders
+ - d3d11deinterlace: deinterlacing filter
+ - d3d11compositor: video composing element
+ - see Windows section below for more details
+
+- new Rust plugins:
+
+ - audiornnoise: Removes noise from an audio stream
+ - awstranscribeparse: Parses AWS audio transcripts into timed text
+ buffers
+ - ccdetect: Detects if valid closed captions are present in a
+ closed captions stream
+ - cea608tojson: Converts CEA-608 Closed Captions to a JSON
+ representation
+ - cmafmux: CMAF fragmented MP4 muxer
+ - dashmp4mux: DASH fragmented MP4 muxer
+ - isofmp4mux: ISO fragmented MP4 muxer
+ - ebur128level: EBU R128 Loudness Level Measurement
+ - ffv1dec: FFV1 video decoder
+ - gtk4paintablesink: GTK4 video sink, which provides a
+ GdkPaintable that can be rendered in various widgets
+ - hlssink3: HTTP Live Streaming sink
+ - hrtfrender: Head-Related Transfer Function (HRTF) renderer
+ - hsvdetector: HSV colorspace detector
+ - hsvfilter: HSV colorspace filter
+ - jsongstenc: Wraps buffers containing any valid top-level JSON
+ structures into higher level JSON objects, and outputs those as
+ ndjson
+ - jsongstparse: Parses ndjson as output by jsongstenc
+ - jsontovtt: converts JSON to WebVTT subtitles
+ - regex: Applies regular expression operations on text
+ - roundedcorners: Adds rounded corners to video
+ - spotifyaudiosrc: Spotify source
+ - textahead: Display upcoming text buffers ahead (e.g. for
+ Karaoke)
+ - transcriberbin: passthrough bin that transcribes raw audio to
+ closed captions using awstranscriber and puts the captions as
+ metas onto the video
+ - tttojson: Converts timed text to a JSON representation
+ - uriplaylistbin: Playlist source bin
+ - webpdec-rs: WebP image decoder with animation support
+
+- New plugin codecalpha with elements to assist with WebM Alpha
+ decoding
+
+ - codecalphademux: Split stream with GstVideoCodecAlphaMeta into
+ two streams
+ - alphacombine: Combine two raw video stream (I420 or NV12) as one
+ stream with alpha channel (A420 or AV12)
+ - vp8alphadecodebin: A bin to handle software decoding of VP8 with
+ alpha
+ - vp9alphadecodebin: A bin to handle software decoding of VP9 with
+ alpha
+
+- New hardware accelerated elements for Linux:
+
+ - v4l2slmpeg2dec: Support for Linux Stateless MPEG2 decoders
+ - v4l2slvp9dec: Support for Linux Stateless VP9 decoders
+ - v4l2slvp8alphadecodebin: Support HW accelerated VP8 with alpha
+ layer decoding
+ - v4l2slvp9alphadecodebin: Support HW accelerated VP9 with alpha
+ layer decoding
New element features and additions
-- this section will be filled in in due course
+- assrender: handle more font mime types; better interaction with
+ matroskademux for embedded fonts
+
+- audiobuffersplit: Add support for specifying output buffer size in
+ bytes (not just duration)
+
+- audiolatency: new "samplesperbuffer" property so users can configure
+ the number of samples per buffer. The default value is 240 samples
+ which is equivalent to 5ms latency with a sample rate of 48000,
+ which might be larger than actual buffer size of audio capture
+ device.
+
+- audiomixer, audiointerleave, GstAudioAggregator: now keep a count of
+ samples that are dropped or processed as statistic and can be made
+ to post QoS messages on the bus whenever samples are dropped by
+ setting the "qos-messages" property on input pads.
+
+- audiomixer, compositor: improved handling of new inputs added at
+ runtime. New API was added to the GstAggregator base class to allow
+ subclasses to opt into an aggregation mode where inactive pads are
+ ignored when processing input buffers
+ (gst_aggregator_set_ignore_inactive_pads(),
+ gst_aggregator_pad_is_inactive()). An “inactive pad” in this context
+ is a pad which, in live mode, hasn’t yet received a first buffer,
+ but has been waited on at least once. What would happen usually in
+ this case is that the aggregator would wait for data on this pad
+ every time, up to the maximum configured latency. This would
+ inadvertently push mixer elements in live mode to the configured
+ latency envelope and delay processing when new inputs are added at
+ runtime until these inputs have actually produced data. This is
+ usually undesirable. With this new API, new inputs can be added
+ (requested) and configured and they won’t delay the data processing.
+ Applications can opt into this new behaviour by setting the
+ "ignore-inactive-pads" property on compositor, audiomixer or other
+ GstAudioAggregator-based elements.
+
+- cccombiner: implement “scheduling” of captions. So far cccombiner’s
+ behaviour was essentially that of a funnel: it strictly looked at
+ input timestamps to associate together video and caption buffers.
+ Now it will try to smoothly schedule caption buffers in order to
+ have exactly one per output video buffer. This might involve
+ rewriting input captions, for example when the input is CDP then
+ sequence counters are rewritten, time codes are dropped and
+ potentially re-injected if the input video frame had a time code
+ meta. This can also lead to the input drifting from synchronization,
+ when there isn’t enough padding in the input stream to catch up. In
+ that case the element will start dropping old caption buffers once
+ the number of buffers in its internal queue reaches a certain limit
+ (configurable via the "max-scheduled" property). The new original
+ funnel-like behaviour can be restored by setting the "scheduling"
+ property to FALSE.
+
+- ccconverter: new "cdp-mode" property to specify which sections to
+ include in CDP packets (timecode, CC data, service info). Various
+ software, including ffmpeg’s Decklink support, fails parsing CDP
+ packets that contain anything but CC data in the CDP packets.
+
+- clocksync: new "sync-to-first" property for automatic timestamp
+ offset setup: if set clocksync will set up the "ts-offset" value
+ based on the first buffer and the pipeline’s running time when the
+ first buffer arrived. The newly configured "ts-offset" in this case
+ would be the value that allows outputting the first buffer without
+ waiting on the clock. This is useful for example to feed a non-live
+ input into an already-running pipeline.
+
+- compositor:
+
+ - multi-threaded input conversion and compositing. Set the
+ "max-threads" property to activate this.
+ - new "sizing-policy" property to support display aspect ratio
+ (DAR)-aware scaling. By default the image is scaled to fill the
+ configured destination rectangle without padding and without
+ keeping the aspect ratio. With sizing-policy=keep-aspect-ratio
+ the input image is scaled to fit the destination rectangle
+ specified by GstCompositorPad:{xpos, ypos, width, height}
+ properties preserving the aspect ratio. As a result, the image
+ will be centered in the destination rectangle with padding if
+ necessary.
+ - new "zero-size-is-unscaled" property on input pads. By default
+ pad width=0 or pad height=0 mean that the stream should not be
+ scaled in that dimension. But if the "zero-size-is-unscaled"
+ property is set to FALSE a width or height of 0 is instead
+ interpreted to mean that the input image on that pad should not
+ be composited, which is useful when creating animations where an
+ input image is made smaller and smaller until it disappears.
+ - improved handling of new inputs at runtime via
+ "ignore-inactive-pads"property (see above for details)
+ - allow output format with alpha even if none of the inputs have
+ alpha (also glvideomixer and other GstVideoAggregator
+ subclasses)
+
+- dashsink: add h265 codec support and signals for allowing custom
+ playlist/fragment output
+
+- decodebin3:
+
+ - improved decoder selection, especially for hardware decoders
+ - make input activation “atomic” when adding inputs dynamically
+ - better interleave handling: take into account decoder latency
+ for interleave size
+
+- decklink:
+
+ - Updated DeckLink SDK to 11.2 to support DeckLink 8K Pro
+ - decklinkvideosrc:
+ - More accurate and stable capture timestamps: use the
+ hardware reference clock time when the frame was finished
+ being captured instead of a clock time much further down the
+ road.
+ - Automatically detect widescreen vs. normal NTSC/PAL
+
+- encodebin:
+
+ - add “smart encoding” support for H.265, VP8 and VP9 (i.e. only
+ re-encode where needed and otherwise pass through encoded video
+ as-is).
+ - H264/H265 smart encoding improvements: respect user-specified
+ stream-format, but if not specified default to avc3/hvc1 with
+ in-band SPS/PPS/VPS signalling for more flexibility.
+ - new encodebin2 element with dynamic/sometimes source pads in
+ order to support the option of doing the muxing outside of
+ encodebin, e.g. in combination with splitmuxsink.
+ - add APIs to set element properties on encoding profiles (see
+ below)
+
+- errorignore: new "ignore-eos" property to also ignore FLOW_EOS from
+ downstream elements
+
+- giosrc: add support for growing source files: applications can
+ specify that the underlying file being read is growing by setting
+ the "is-growing" property. If set, the source won’t EOS when it
+ reaches the end of the file, but will instead start monitoring it
+ and will start reading data again whenever a change is detected. The
+ new "waiting-data" and "done-waiting-data" signals keep the
+ application informed about the current state.
+
+- gtksink, gtkglsink:
+
+ - scroll event support: forwarded as navigation events into the
+ pipeline
+ - "video-aspect-ratio-override" property to force a specific
+ aspect ratio
+ - "rotate-method" property and support automatic rotation based on
+ image tags
+
+- identity: new "stats" property allows applications to retrieve the
+ number of bytes and buffers that have passed through so far.
+
+- interlace: add support for more formats, esp 10-bit, 12-bit and
+ 16-bit ones
+
+- jack: new "low-latency" property for automatic latency-optimized
+ setting and "port-names" property to select ports explicitly
+
+- jpegdec: support output conversion to RGB using libjpeg-turbo (for
+ certain input files)
+
+- line21dec:
+
+ - "mode" property to control whether and how detected closed
+ captions should be inserted in the list of existing close
+ caption metas on the input frame (if any): add, drop, or
+ replace.
+ - "ntsc-only" property to only look for captions if video has NTSC
+ resolution
+
+- line21enc: new "remove-caption-meta" to remove metas from output
+ buffers after encoding the captions into the video data; support for
+ CDP closed captions
+
+- matroskademux, matroskamux: Add support for ffv1, a lossless
+ intra-frame video coding format.
+
+- matroskamux: accept in-band SPS/PPS/VPS for H264 and H265
+ (i.e. stream-format avc3 and hev1) which allows on-the-fly
+ profile/level/resolution changes.
+
+- matroskamux: new "cluster-timestamp-offset" property, useful for use
+ cases where the container timestamps should map to some absolute
+ wall clock time, for example.
+
+- rtpsrc: add "caps" property to allow explicit setting of the caps
+ where needed
+
+- mpegts: support SCTE-35 passthrough via new "send-scte35-events"
+ property on MPEG-TS demuxer tsdemux. When enabled, SCTE 35 sections
+ (eg ad placement opportunities) are forwarded as events donwstream
+ where they can be picked up again by mpegtsmux. This required a
+ semantic change in the SCTE-35 section API: timestamps are now in
+ running time instead of muxer pts.
+
+- tsdemux: Handle PCR-less MPEG-TS streams; more robust timestamp
+ handling in certain corner cases and for poorly muxed streams.
+
+- mpegtsmux:
+
+ - More conformance improvements to make MPEG-TS analyzers happy:
+ - PCR timing accuracy: Improvements to the way mpegtsmux
+ outputs PCR observations in CBR mode, so that a PCR
+ observation is always inserted when needed, so that we never
+ miss the configured pcr-interval, as that triggers various
+ MPEG-TS analyser errors.
+ - Improved PCR/SI scheduling
+ - Don’t write PCR until PAT/PMT are output to make sure streams
+ start cleanly with a PAT/PMT.
+ - Allow overriding the automatic PMT PID selection via
+ application-supplied PMT_%d fields in the prog-map
+ structure/property.
+
+- mp4mux:
+
+ - new "first-moov-then-finalise" mode for fragmented output where
+ the output will start with a self-contained moov atom for the
+ first fragment, and then produce regular fragments. Then at the
+ end when the file is finalised, the initial moov is invalidated
+ and a new moov is written covering the entire file. This way the
+ file is a “fragmented mp4” file while it is still being written
+ out, and remains playable at all times, but at the end it is
+ turned into a regular mp4 file (with former fragment headers
+ remaining as unused junk data in the file).
+ - support H.264 avc3 and H.265 hvc1 stream formats as input where
+ the codec data is signalled in-band inside the bitstream instead
+ of caps/file headers.
+ - support profile/level/resolution changes for H264/H265 input
+ streams (i.e. codec data changing on the fly). Each codec_data
+ is put into its own SampleTableEntry inside the stsd, unless the
+ input is in avc3 stream format in which case it’s written
+ in-band an not in the headers.
+
+- multifilesink: new ""min-keyframe-distance"" property to make
+ minimum distance between keyframes in next-file=key-frame mode
+ configurable instead of hard-coding it to 10 seconds.
+
+- mxfdemux has seen a big refactoring to support non-frame wrappings
+ and more accurate timestamp/seek handling for some formats
+
+- msdk plugin for hardware-accelerated video encoding and decoding
+ using the Intel Media SDK:
+
+ - oneVPL support (Intel oneAPI Video Processing Library)
+ - AV1 decoding support
+ - H.264 decoder now supports constrained-high and progressive-high
+ profiles
+ - H.264 encoder:
+ - more configuration options (properties):
+ "intra-refresh-type", "min-qp" , "max-qp", "p-pyramid",
+ "dblk-idc"
+ - H.265 encoder:
+ - can output main-still-picture profile
+ - now inserts HDR SEIs (mastering display colour volume and
+ content light level)
+ - more configuration options (properties):
+ "intra-refresh-type", "min-qp" , "max-qp", "p-pyramid",
+ "b-pyramid", "dblk-idc", "transform-skip"
+ - support for RGB 10bit format
+ - External bitrate control in encoders
+ - Video post proc element msdkvpp gained support for 12-bit pixel
+ formats P012_LE, Y212_LE and Y412_LE
+
+- nvh264sldec: interlaced stream support
+
+- openh264enc: support main, high, constrained-high and
+ progressive-high profiles
+
+- openjpeg: support for multithreaded decoding and encoding
+
+- rtspsrc: now supports IPv6 also for tunneled mode (RTSP-over-HTTP);
+ new "ignore-x-server-reply" property to ignore the
+ x-server-ip-address server header reply in case of HTTP tunneling,
+ as it is often broken.
+
+- souphttpsrc: Runtime compatibility support for libsoup2 and
+ libsoup3. libsoup3 is the latest major version of libsoup, but
+ libsoup2 and libsoup3 can’t co-exist in the same process because
+ there is no namespacing or versioning for GObject types. As a
+ result, it would be awkward if the GStreamer souphttpsrc plugin
+ linked to a specific version of libsoup, because it would only work
+ with applications that use the same version of libsoup. To make this
+ work, the soup plugin now tries to determine the libsoup version
+ used by the application (and its other dependencies) at runtime on
+ systems where GStreamer is linked dynamically. libsoup3 support is
+ still considered somewhat experimental at this point.
+
+- srtsrc, srtsink: add signals for the application to accept/reject
+ incoming connections
+
+- timeoverlay: new elapsed-running-time time mode which shows the
+ running time since the first running time (and each flush-stop).
+
+- udpsrc: new timestamping mode to retrieve packet receive timestamps
+ from the kernel via socket control messages (SO_TIMESTAMPNS) on
+ supported platforms
+
+- uritranscodebin: new setup-source and element-setup signals for
+ applications to configure elements used
+
+- v4l2codecs plugin gained support for 4x4 and 32x32 tile formats
+ enabling some platforms or direct renders. Important memory usage
+ improvement.
+
+- v4l2slh264dec now implements the final Linux uAPI as shipped on
+ Linux 5.11 and later.
+
+- valve: add "drop-mode" property and provide two new modes of
+ operation: in drop-mode=forward-sticky-events sticky events
+ (stream-start, segment, tags, caps, etc.) are forwarded downstream
+ even when dropping is enabled; drop-mode=transform-to-gap will in
+ addition also convert buffers into gap events when dropping is
+ enabled, which lets downstream elements know that time is advancing
+ and might allow for preroll in many scenarios. By default all events
+ and all buffers are dropped when dropping is enabled, which can
+ cause problems with caps negotiation not progressing or branches not
+ prerolling when dropping is enabled.
+
+- videocrop: support for many more pixel formats, e.g. planar YUV
+ formats with > 8bits and GBR* video formats; can now also accept
+ video not backed by system memory as long as downstream supports the
+ GstCropMeta
+
+- videotestsrc: new smpte-rp-219 pattern for SMPTE75 RP-219 conformant
+ color bars
+
+- vp8enc: finish support for temporal scalability: two new properties
+ ("temporal-scalability-layer-flags",
+ "temporal-scalability-layer-sync-flags") and a unit change on the
+ "temporal-scalability-target-bitrate" property (now expects bps);
+ also make temporal scalability details available to RTP payloaders
+ as buffer metadata.
+
+- vp9enc: new properties to tweak encoder performance:
+
+ - "aq-mode" to configure adaptive quantization modes
+ - "frame-parallel-decoding" to configure whether to create a
+ bitstream that reduces decoding dependencies between frames
+ which allows staged parallel processing of more than one video
+ frames in the decoder. (Defaults to TRUE)
+ - "row-mt", "tile-columns" and "tile-rows" so multithreading can
+ be enabled on a per-tile basis, instead of on a per tile-column
+ basis. In combination with the new "tile-rows" property, this
+ allows the encoder to make much better use of the available CPU
+ power.
+
+- vp9dec, vp9enc: add support for 10-bit 4:2:0 and 4:2:2 YUV, as well
+ as 8-bit 4:4:4
+
+- vp8enc, vp9enc now default to “good quality” for the deadline
+ property rather then “best quality”. Having the deadline set to best
+ quality causes the encoder to be absurdly slow, most real-life users
+ will prefer good-enough quality with better performance instead.
+
+- wpesrc:
+
+ - implement audio support: a new sometimes source pad will be
+ created for each audio stream created by the web engine.
+ - move wpesrc to wpevideosrc and add a wrapper bin wpesrc to also
+ support audio
+ - also handles web:// URIs now (same as cefsrc)
+ - post messages with the estimated load progress on the bus
-Plugin and library moves
+- x265enc: add negative DTS support, which means timestamps are now
+ offset by 1h same as with x264enc
+
+RTP Payloaders and Depayloaders
+
+- rtpisacpay, rtpisacdepay: new RTP payloader and depayloader for iSAC
+ audio codec
+
+- rtph264depay:
+
+ - new "request-keyframe" property to make the depayloader
+ automatically request a new keyframe from the sender on packet
+ loss, consistent with the new property on rtpvp8depay.
+ - new "wait-for-keyframe" property to make depayloader wait for a
+ new keyframe at the beginning and after packet loss (only
+ effective if the depayloader outputs AUs), consistent with the
+ existing property on rtpvp8depay.
+
+- rtpopuspay, rtpopusdepay: support libwebrtc-compatible multichannel
+ audio in addition to the previously supported multichannel audio
+ modes
+
+- rtpopuspay: add DTX (Discontinuous Transmission) support
+
+- rtpvp8depay: new "request-keyframe" property to make the depayloader
+ automatically request a new keyframe from the sender on packet loss.
+
+- rtpvp8pay: temporal scaling support
-- this section will be filled in in due course
+- rtpvp9depay: Improved SVC handling (aggregate all layers)
+
+RTP Infrastructure
+
+- rtpst2022-1-fecdec, rtpst2022-1-fecenc: new elements providing SMPTE
+ 2022-1 2-D Forward Error Correction. More details in Mathieu’s blog
+ post.
+
+- rtpreddec: BUNDLE support
+
+- rtpredenc, rtpulpfecenc: add support for Transport-wide Congestion
+ Control (TWCC)
+
+- rtpsession: new "twcc-feedback-interval" property to allow RTCP TWCC
+ reports to be scheduled on a timer instead of per marker-bit.
+
+Plugin and library moves
- There were no plugin moves or library moves in this cycle.
The following elements or plugins have been removed:
-- this section will be filled in in due course
-
- The ofa audio fingerprinting plugin has been removed. The MusicIP
database has been defunct for years so this plugin is likely neither
useful nor used by anyone.
Miscellaneous API additions
-- this section will be filled in in due course
+Core
+
+- gst_buffer_new_memdup() is a convenience function for the
+ widely-used gst_buffer_new_wrapped(g_memdup(data,size),size)
+ pattern.
+
+- gst_caps_features_new_single() creates a new single GstCapsFeatures,
+ avoiding the need to use the vararg function with NULL terminator
+ for simple cases.
+
+- gst_element_type_set_skip_documentation() can be used by plugins to
+ signal that certain elements should not be included in the GStreamer
+ plugin documentation. This is useful for plugins where elements are
+ registered dynamically based on hardware capabilities and/or where
+ the available plugins and properties vary from system to system.
+ This is used in the d3d11 plugin for example to ensure that only the
+ list of default elements is advertised in the documentation.
+
+- gst_type_find_suggest_empty_simple() is a new convenience function
+ for typefinders for cases where there’s only a media type and no
+ other fields.
+
+- New API to create elements and set properties at construction time,
+ which is not only convenient, but also allows GStreamer elements to
+ have construct-only properties: gst_element_factory_make_full(),
+ gst_element_factory_make_valist(),
+ gst_element_factory_make_with_properties(),
+ gst_element_factory_create_full(),
+ gst_element_factory_create_valist(),
+ gst_element_factory_create_with_properties().
+
+- GstSharedTaskPool: new “shared” task pool subclass with slightly
+ different default behaviour than the existing GstTaskPool which
+ would create unlimited number of threads for new tasks. The shared
+ taskpool creates up to N threads (default: 1) and then distributes
+ pending tasks to those threads round-robin style, and blocks if no
+ thread is available. It is possible to join tasks. This can be used
+ by plugins to implement simple multi-threaded processing and is used
+ for the new multi-threaded video conversion and compositing done in
+ GstVideoAggregator, videoconverter and compositor.
+
+Plugins Base Utils library
+
+- GstDiscoverer:
+
+ - gst_discoverer_container_info_get_tags() was added to retrieve
+ global/container tags (vs. per-stream tags). Per-Stream tags can
+ be retrieved via the existing
+ gst_discoverer_stream_info_get_tags().
+ gst_discoverer_info_get_tags(), which for many files returns a
+ confusing mix of stream and container tags, has been deprecated
+ in favour of the container/stream-specific functions.
+ - gst_discoverer_stream_info_get_stream_number() returns a unique
+ integer identifier for a given stream within the given
+ GstDiscoverer context. (If this matches the stream number inside
+ the container bitstream that’s by coincidence and not by
+ design.)
+
+- gst_pb_utils_get_caps_description_flags() can be used to query
+ whether certain caps represent a container, audio, video, image,
+ subtitles, tags, or something else. This only works for formats
+ known to GStreamer.
+
+- gst_pb_utils_get_file_extension_from_caps() returns a possible file
+ extension for given caps.
+
+- gst_codec_utils_h264_get_profile_flags_level(): Parses profile,
+ flags, and level from H264 AvcC codec_data. The format of H264 AVCC
+ extradata/sequence_header is documented in the ITU-T H.264
+ specification section 7.3.2.1.1 as well as in ISO/IEC 14496-15
+ section 5.3.3.1.2.
+
+- gst_codec_utils_caps_get_mime_codec() to convert caps to a RFC 6381
+ compatible MIME codec string codec. Useful for providing the codecs
+ field inside the Content-Type HTTP header for containerized formats,
+ such as mp4 or matroska.
+
+GStreamer OpenGL integration library and plugins
+
+- glcolorconvert: added suppport for converting the video formats
+ A420, AV12, BGR, BGRA, RGBP and BGRP.
+
+- Added support to GstGLBuffer for persistent buffer mappings where a
+ Pixel Buffer Object (PBO) can be mapped by both the CPU and the GPU.
+ This removes a memcpy() when uploading textures or vertices
+ particularly when software decoders (e.g. libav) are direct
+ rendering into our memory. Improves transfer performance
+ significantly. Requires OpenGL 4.4, GL_ARB_buffer_storage or
+ GL_EXT_buffer_storage
+
+- Added various helper functions for handling 4x4 matrices of affine
+ transformations as used by GstVideoAffineTransformationMeta.
+
+- Add support to GstGLContext for allowing the application to control
+ the config (EGLConfig, GLXConfig, etc) used when creating the OpenGL
+ context. This allows the ability to choose between RGB16 or RGB10A2
+ or RGBA8 back/front buffer configurations that were previously
+ hardcoded. GstGLContext also supports retrieving the configuration
+ it was created with or from an externally provide OpenGL context
+ handle. This infrastructure is also used to create a compatible
+ config from an application/externally provided OpenGL context in
+ order to improve compatibility with other OpenGL frameworks and GUI
+ toolkits. A new environment variable GST_GL_CONFIG was also added to
+ be able to request a specific configuration from the command line.
+ Note: different platforms will have different functionality
+ available.
+
+- Add support for choosing between EGL and WGL at runtime when running
+ on Windows. Previously this was a build-time switch. Allows use in
+ e.g. Gtk applications on Windows that target EGL/ANGLE without
+ recompiling GStreamer. gst_gl_display_new_with_type() can be used by
+ applications to choose a specific display type to use.
+
+- Build fixes to explicitly check for Broadcom-specific libraries on
+ older versions of the Raspberry Pi platform. The Broadcom OpenGL ES
+ and EGL libraries have different filenames. Using the vc4 Mesa
+ driver on the Raspberry Pi is not affected.
+
+- Added support to glupload and gldownload for transferring RGBA
+ buffers using the memory:NVMM available on the Nvidia Tegra family
+ of embedded devices.
+
+- Added support for choosing libOpenGL and libGLX as used in a GLVND
+ environment on unix-based platforms. This allows using desktop
+ OpenGL and EGL without pulling in any GLX symbols as would be
+ required with libGL.
+
+Video library
+
+- New raw video formats:
+
+ - AV12 (NV12 with alpha plane)
+ - RGBP and BGRP (planar RGB formats)
+ - ARGB64 variants with specified endianness instead of host
+ endianness:
+ - ARGB64_LE, ARGB64_BE
+ - RGBA64_BE, RGBA64_LE
+ - BGRA64_BE, BGRA64_LE
+ - ABGR64_BE, ABGR64_LE
+
+- gst_video_orientation_from_tag() is new convenience API to parse the
+ image orientation from a GstTagList.
+
+- GstVideoDecoder subframe support (see below)
+
+- GstVideoCodecState now also carries some HDR metadata
+
+- Ancillary video data: implement transform functions for AFD/Bar
+ metas, so they will be forwarded in more cases
+
+MPEG-TS library
+
+This library only handles section parsing and such, see above for
+changes to the actual mpegtsmux and mpegtsdemux elements.
+
+- many additions and improvements to SCTE-35 section parsing
+- new API for fetching extended descriptors:
+ gst_mpegts_find_descriptor_with_extension()
+- add support for SIT sections (Selection Information Tables)
+- expose event-from-section constructor gst_event_new_mpegts_section()
+- parse Audio Preselection Descriptor needed for Dolby AC-4
+
+GstWebRTC library + webrtcbin
+
+- Change the way in which sink pads and transceivers are matched
+ together to support easier usage. If a pad is created without a
+ specific index (i.e. using sink_%u as the pad template), then an
+ available compatible transceiver will be searched for. If a specific
+ index is requested (i.e. sink_1) then if a transceiver for that
+ m-line already exists, that transceiver must match the new sink pad
+ request. If there is no transceiver available in either scenario, a
+ new transceiver is created. If a mixture of both sink_1 and sink_%u
+ requests result in an impossible situation, an error will be
+ produced at pad request time or from create offer/answer.
+
+- webrtcbin now uses regular ICE nomination instead of libnice’s
+ default of aggressive ICE nomination. Regular ICE nomination is the
+ default recommended by various relevant standards and improves
+ connectivity in specific network scenarios.
+
+- Add support for limiting the port range used for RTP with the
+ addition of the min-rtp-port and max-rtp-port properties on the ICE
+ object.
+
+- Expose the SCTP transport as a property on webrtcbin to more closely
+ match the WebRTC specification.
+
+- Added support for taking into account the data channel transport
+ state when determining the value of the "connection-state" property.
+ Previous versions of the WebRTC spec did not include the data
+ channel state when computing this value.
+
+- Add configuration for choosing the size of the underlying sockets
+ used for transporting media data
+
+- Always advertise support for the transport-cc RTCP feedback protocol
+ as rtpbin supports it. For full support, the configured caps (input
+ or through codec-preferences) need to include the relevant RTP
+ header extension.
+
+- Numerous fixes to caps and media handling to fail-fast when an
+ incompatible situation is detected.
+
+- Improved support for attaching the required media after a remote
+ offer has been set.
+
+- Add support for dynamically changing the amount of FEC used for a
+ particular stream.
+
+- webrtcbin now stops further SDP processing at the first error it
+ encounters.
+
+- Completed support for either local or the remote closing a data
+ channel.
+
+- Various fixes when performing BUNDLEing of the media streams in
+ relation to RTX and FEC usage.
+
+- Add support for writing out QoS DSCP marking on outgoing packets to
+ improve reliability in some network scenarios.
+
+- Improvements to the statistics returned by the get-stats signal
+ including the addition of the raw statistics from the internal
+ RTPSource, the TWCC stats when available.
+
+- The webrtc library does not expose any objects anymore with public
+ fields. Instead properties have been added to replace that
+ functionality. If you are accessing such fields in your application,
+ switch to the corresponding properties.
+
+GstCodecs and Video Parsers
+
+- Support for render delays to improve throughput across all CODECs
+ (used with NVDEC and V4L2).
+- lots of improvements to parsers and the codec parsing decoder base
+ classes (H264, H265, VP8, VP9, AV1, MPEG-2) used for various
+ hardware-accelerate decoder APIs.
+
+Bindings support
+
+- gst_allocation_params_new() allocates a GstAllocationParams struct
+ on the heap. This should only be used by bindings (and freed via
+ gst_allocation_params_free() then). In C code you would allocate
+ this on the stack and only init it in place.
+
+- gst_debug_log_literal() can be used to log a string to the debug log
+ without going through any printf format expansion and associated
+ overhead. This is mostly useful for bindings such as the Rust
+ bindings which may have done their own formatting already .
+
+- Provide non-inlined versions of refcounting APIs for various
+ GStreamer mini objects, so that they can be consumed by bindings
+ (e.g. gstreamer-sharp): gst_buffer_ref, gst_buffer_unref,
+ gst_clear_buffer, gst_buffer_copy, gst_buffer_replace,
+ gst_buffer_list_ref, gst_buffer_list_unref, gst_clear_buffer_list,
+ gst_buffer_list_copy, gst_buffer_list_replace, gst_buffer_list_take,
+ gst_caps_ref, gst_caps_unref, gst_clear_caps, gst_caps_replace,
+ gst_caps_take, gst_context_ref, gst_context_unref, gst_context_copy,
+ gst_context_replace, gst_event_replace, gst_event_steal,
+ gst_event_take, gst_event_ref, gst_event_unref, gst_clear_event,
+ gst_event_copy, gst_memory_ref, gst_memory_unref, gst_message_ref,
+ gst_message_unref, gst_clear_message, gst_message_copy,
+ gst_message_replace, gst_message_take, gst_promise_ref,
+ gst_promise_unref, gst_query_ref, gst_query_unref, gst_clear_query,
+ gst_query_copy, gst_query_replace, gst_query_take, gst_sample_ref,
+ gst_sample_unref, gst_sample_copy, gst_tag_list_ref,
+ gst_tag_list_unref, gst_clear_tag_list, gst_tag_list_replace,
+ gst_tag_list_take, gst_uri_copy, gst_uri_ref, gst_uri_unref,
+ gst_clear_uri.
+
+- expose a GType for GstMiniObject
+
+- gst_device_provider_probe() now returns non-floating device object
+
+API Deprecations
+
+- gst_element_get_request_pad() has been deprecated in favour of the
+ newly-added gst_element_request_pad_simple() which does the exact
+ same thing but has a less confusing name that hopefully makes clear
+ that the function request a new pad rather than just retrieves an
+ already-existing request pad.
+
+- gst_discoverer_info_get_tags(), which for many files returns a
+ confusing mix of stream and container tags, has been deprecated in
+ favour of the container-specific and stream-specific functions,
+ gst_discoverer_container_info_get_tags() and
+ gst_discoverer_stream_info_get_tags().
+
+- gst_video_sink_center_rect() was deprecated in favour of the more
+ generic newly-added gst_video_center_rect().
+
+- The GST_MEMORY_FLAG_NO_SHARE flag has been deprecated, as it tends
+ to cause problems and prevents sub-buffering. If pooling or lifetime
+ tracking is required, memories should be allocated through a custom
+ GstAllocator instead of relying on the lifetime of the buffers the
+ memories were originally attached to, which is fragile anyway.
+
+- The GstPlayer high-level playback library is being replaced with the
+ new GstPlay library (see above). GstPlayer should be considered
+ deprecated at this point and will be marked as such in the next
+ development cycle. Applications should be ported to GstPlay.
+
+- Gstreamer Editing Services: ges_video_transition_set_border(),
+ ges_video_transition_get_border()
+ ges_video_transition_set_inverted()
+ ges_video_transition_is_inverted() have been deprecated, use
+ ges_timeline_element_set_children_properties() instead.
Miscellaneous performance, latency and memory optimisations
-- this section will be filled in in due course
+More video conversion fast paths
-Miscellaneous other changes and enhancements
+- v210 ↔ I420, YV12, Y42B, UYVY and YUY2
+- A420 → RGB
+
+Less jitter when waiting on the system clock
+
+- Better system clock wait accuracy, less jitter: where available,
+ clock_nanosleep is used for higher accuracy for waits below 500
+ usecs, and waits below 2ms will first use the regular waiting system
+ and then clock_nanosleep for the remainder. The various wait
+ implementation have a latency ranging from 50 to 500+ microseconds.
+ While this is not a major issue when dealing with a low number of
+ waits per second (for ex: video), it does introduce a non-negligible
+ jitter for synchronisation of higher packet rate systems.
+
+Video decoder subframe support
-- this section will be filled in in due course
+- The GstVideoDecoder base class gained API to process input at the
+ sub-frame level. That way video decoders can start decoding slices
+ before they have received the full input frame in its entirety (to
+ the extent this is supported by the codec, of course). This helps
+ with CPU utilisation and reduces latency.
-Tracing framework and debugging improvements
+- This functionality is now being used in the OpenJPEG JPEG 2000
+ decoder, the FFmpeg H.264 decoder (in case of NAL-aligned input) and
+ the OpenMAX H.264/H.265 decoders (in case of NAL-aligned input).
-- this section will be filled in in due course
+Miscellaneous other changes and enhancements
+
+- GstDeviceMonitor no longer fails to start just because one of the
+ device providers failed to start. That could happen for example on
+ systems where the pulseaudio device provider is installed, but
+ pulseaudio isn’t actually running but ALSA is used for audio
+ instead. In the same vein the device monitor now keeps track of
+ which providers have been started (via the new
+ gst_device_provider_is_started()) and only stops actually running
+ device providers when stopping the device monitor.
+
+- On embedded systems it can be useful to create a registry that can
+ be shared and read by multiple processes running as different users.
+ It is now possible to set the new GST_REGISTRY_MODE environment
+ variable to specify the file mode for the registry file, which by
+ default is set to be only user readable/writable.
+
+- GstNetClientClock will signal lost sync in case the remote time
+ resets (e.g. because device power cycles), by emitting the “synced”
+ signal with synced=FALSE parameter, so applications can take action.
+
+- gst_value_deserialize_with_pspec() allows deserialization with a
+ hint for what the target GType should be. This allows for example
+ passing arrays of flags through the command line or
+ gst_util_set_object_arg(), eg: foo="<bar,bar+baz>".
+
+- It’s now allowed to create an empty GstVideoOverlayComposition
+ without any rectangles by passing a NULL rectangle to
+ gst_video_overlay_composition_new(). This is useful for bindings and
+ simplifies application code in some places.
+
+Tracing framework, debugging and testing improvements
+
+- New factories tracer to list loaded elements (and other plugin
+ features). This can be useful to collect a list of elements needed
+ for an application, which then in turn can be used to create a
+ tailored minimal GStreamer build that contains just the elements
+ needed and nothing else.
+- New plugin-feature-loaded tracing hook for use by tracers like the
+ new factories tracer
+
+- GstHarness: Add gst_harness_set_live() so that harnesses can be set
+ to non-live and return is-live=false in latency queries if needed.
+ Default behaviour is to always return is-live=true in latency
+ queries.
+
+- navseek: new "hold-eos" property. When enabled, the element will
+ hold back an EOS event until the next keystroke (via navigation
+ events). This can be used to keep a video sink showing the last
+ frame of a video pipeline until a key is pressed instead of tearing
+ it down immediately on EOS.
+
+- New fakeaudiosink element: mimics an audio sink and can be used for
+ testing and CI pipelines on systems where no audio system is
+ installed or running. It differs from fakesink in that it only
+ support audio caps and syncs to the clock by default like a normal
+ audio sink. It also implements the GstStreamVolume interface like
+ most audio sinks do.
+
+- New videocodectestsink element for video codec conformance testing:
+ Calculates MD5 checksums for video frames and skips any padding
+ whilst doing so. Can optionally also write back the video data with
+ padding removed into a file for easy byte-by-byte comparison with
+ reference data.
Tools
-- this section will be filled in in due course
+gst-inspect-1.0
+
+- Can sort the list of plugins by passing --sort=name as command line
+ option
+
+gst-launch-1.0
+
+- will now error out on top-level properties that don’t exist and
+ which were silently ignored before
+- On Windows the high-resolution clock is enabled now, which provides
+ better clock and timer performance on Windows (see Windows section
+ below for more details).
+
+gst-play-1.0
+
+- New --start-position command line argument to start playback from
+ the specified position
+- Audio can be muted/unmuted in interactive mode by pressing the m
+ key.
+- On Windows the high-resolution clock is enabled now (see Windows
+ section below for more details)
+
+gst-device-monitor-1.0
+
+- New --include-hidden command line argument to also show “hidden”
+ device providers
+
+ges-launch-1.0
+
+- New interactive mode that allows seeking and such. Can be disabled
+ by passing the --no-interactive argument on the command line.
+- Option to forward tags
+- Allow using an existing clip to determine the rendering format (both
+ topology and profile) via new --profile-from command line argument.
GStreamer RTSP server
-- this section will be filled in in due course
+- GstRTSPMediaFactory gained API to disable RTCP
+ (gst_rtsp_media_factory_set_enable_rtcp(), "enable-rtcp" property).
+ Previously RTCP was always allowed for all RTSP medias. With this
+ change it is possible to disable RTCP completely, no matter if the
+ client wants to do RTCP or not.
+
+- Make a mount point of / work correctly. While not allowed by the
+ RTSP 2 spec, the RTSP 1 spec is silent on this and it is used in the
+ wild. It is now possible to use / as a mount path in
+ gst-rtsp-server, e.g. rtsp://example.com/ would work with this now.
+ Note that query/fragment parts of the URI are not necessarily
+ correctly handled, and behaviour will differ between various
+ client/server implementations; so use it if you must but don’t bug
+ us if it doesn’t work with third party clients as you’d hoped.
+
+- multithreading fixes (races, refcounting issues, deadlocks)
+
+- ONVIF audio backchannel fixes
+
+- ONVIF trick mode optimisations
+
+- rtspclientsink: new "update-sdp" signal that allows updating the SDP
+ before sending it to the server via ANNOUNCE. This can be used to
+ add additional metadata to the SDP, for example. The order and
+ number of medias must not be changed, however.
GStreamer VAAPI
-- this section will be filled in in due course
+- new AV1 decoder element (vaapiav1dec)
+
+- H264 decoder: handle stereoscopic 3D video with frame packing
+ arrangement SEI messages
+
+- H265 encoder: added Screen Content Coding extensions support
+
+- H265 decoder: gained MAIN_444_12 profile support (decoded to
+ Y412_LE), and 4:2:2 12-bits support (decoded to Y212_LE)
+
+- vaapipostproc: gained BT2020 color standard support
+
+- vaapidecode: now generates caps templates dynamically at runtime in
+ order to advertise actually supported caps instead of all
+ theoretically supported caps.
+
+- GST_VAAPI_DRM_DEVICE environment variable to force a specified DRM
+ device when a DRM display is used. It is ignored when other types of
+ displays are used. By default /dev/dri/renderD128 is used for DRM
+ display.
GStreamer OMX
-- this section will be filled in in due course
+- subframe support in H.264/H.265 decoders
GStreamer Editing Services and NLE
-- this section will be filled in in due course
+- framepositioner: new "operator" property to access blending modes in
+ the compositor
+- timeline: Implement snapping to markers
+- smart-mixer: Add support for d3d11compositor and glvideomixer
+- titleclip: add "draw-shadow" child property
+- ges:// URI support to define a timeline from a description.
+- command-line-formatter
+ - Add track management to timeline description
+ - Add keyframe support
+- ges-launch-1.0:
+ - Add an interactive mode where we can seek etc…
+ - Add option to forward tags
+ - Allow using an existing clip to determine the rendering format
+ (both topology and profile) via new --profile-from command line
+ argument.
+- Fix static build
GStreamer validate
-- this section will be filled in in due course
+- report: Add a way to force backtraces on reports even if not a
+ critical issue (GST_VALIDATE_ISSUE_FLAGS_FORCE_BACKTRACE)
+- Add a flag to gst_validate_replace_variables_in_string() allow
+ defining how to resolve variables in structs
+- Add gst_validate_bin_monitor_get_scenario() to get the bin monitor
+ scenario, which is useful for applications that use Validate
+ directly.
+- Add an expected-values parameter to wait, message-type=XX allowing
+ more precise filtering of the message we are waiting for.
+- Add config file support: each test can now use a config file for the
+ given media file used to test.
+- Add support to check properties of object properties
+- scenario: Add an "action-done" signal to signal when an action is
+ done
+- scenario: Add a "run-command" action type
+- scenario: Allow forcing running action on idle from scenario file
+- scenario: Allow iterating over arrays in foreach
+- scenario: Rename ‘interlaced’ action to ‘non-blocking’
+- scenario: Add a non-blocking flag to the wait signal
GStreamer Python Bindings
-- this section will be filled in in due course
+- Fixes for Python 3.10
+- Various build fixes
+- at least one known breaking change caused by g-i annotation changes
+ (see below)
GStreamer C# Bindings
-- this section will be filled in in due course
+- Fix GstDebugGraphDetails enum
+- Updated to latests GtkSharp
+- Updated to include GStreamer 1.20 API
GStreamer Rust Bindings and Rust Plugins
-The GStreamer Rust bindings are released separately with a different
-release cadence that’s tied to gtk-rs, but the latest release has
-already been updated for the upcoming new GStreamer 1.20 API.
-
-gst-plugins-rs, the module containing GStreamer plugins written in Rust,
-has also seen lots of activity with many new elements and plugins.
-
-What follows is a list of elements and plugins available in
-gst-plugins-rs, so people don’t miss out on all those potentially useful
-elements that have no C equivalent.
-
-- FIXME: add new elements
-
-Rust audio plugins
-
-- audiornnoise: New element for audio denoising which implements the
- noise removal algorithm of the Xiph RNNoise library, in Rust
-- rsaudioecho: Port of the audioecho element from gst-plugins-good
- rsaudioloudnorm: Live audio loudness normalization element based on
- the FFmpeg af_loudnorm filter
-- claxondec: FLAC lossless audio codec decoder element based on the
- pure-Rust claxon implementation
-- csoundfilter: Audio filter that can use any filter defined via the
- Csound audio programming language
-- lewtondec: Vorbis audio decoder element based on the pure-Rust
- lewton implementation
-
-Rust video plugins
-
-- cdgdec/cdgparse: Decoder and parser for the CD+G video codec based
- on a pure-Rust CD+G implementation, used for example by karaoke CDs
-- cea608overlay: CEA-608 Closed Captions overlay element
-- cea608tott: CEA-608 Closed Captions to timed-text (e.g. VTT or SRT
- subtitles) converter
-- tttocea608: CEA-608 Closed Captions from timed-text converter
-- mccenc/mccparse: MacCaption Closed Caption format encoder and parser
-- sccenc/sccparse: Scenarist Closed Caption format encoder and parser
-- dav1dec: AV1 video decoder based on the dav1d decoder implementation
- by the VLC project
-- rav1enc: AV1 video encoder based on the fast and pure-Rust rav1e
- encoder implementation
-- rsflvdemux: Alternative to the flvdemux FLV demuxer element from
- gst-plugins-good, not feature-equivalent yet
-- rsgifenc/rspngenc: GIF/PNG encoder elements based on the pure-Rust
- implementations by the image-rs project
-
-Rust text plugins
-
-- textwrap: Element for line-wrapping timed text (e.g. subtitles) for
- better screen-fitting, including hyphenation support for some
- languages
-
-Rust network plugins
-
-- reqwesthttpsrc: HTTP(S) source element based on the Rust
- reqwest/hyper HTTP implementations and almost feature-equivalent
- with the main GStreamer HTTP source souphttpsrc
-- s3src/s3sink: Source/sink element for the Amazon S3 cloud storage
-- awstranscriber: Live audio to timed text transcription element using
- the Amazon AWS Transcribe API
-
-Generic Rust plugins
-
-- sodiumencrypter/sodiumdecrypter: Encryption/decryption element based
- on libsodium/NaCl
-- togglerecord: Recording element that allows to pause/resume
- recordings easily and considers keyframe boundaries
-- fallbackswitch/fallbacksrc: Elements for handling potentially
- failing (network) sources, restarting them on errors/timeout and
- showing a fallback stream instead
-- threadshare: Set of elements that provide alternatives for various
- existing GStreamer elements but allow to share the streaming threads
- between each other to reduce the number of threads
-- rsfilesrc/rsfilesink: File source/sink elements as replacements for
- the existing filesrc/filesink elements
+- The GStreamer Rust bindings are released separately with a different
+ release cadence that’s tied to gtk-rs, but the latest release has
+ already been updated for the upcoming new GStreamer 1.20 API (v1_20
+ feature).
+
+- gst-plugins-rs, the module containing GStreamer plugins written in
+ Rust, has also seen lots of activity with many new elements and
+ plugins. See the New Elements section above for a list of new Rust
+ elements.
Build and Dependencies
- Meson 0.59 or newer is required to build GStreamer now.
-- FIXME: this section will be filled in in due course
+- The GLib requirement has been bumped to GLib 2.56 or newer (from
+ March 2018).
+
+- The wpe plugin now requires wpe >= 2.28 and wpebackend-fdo >= 1.8
Explicit opt-in required for build of certain plugins with (A)GPL dependencies
See Building plugins with (A)GPL-licensed dependencies for more details
and a non-exhaustive list of plugins affected.
-gst-build: replaced by Monorepo
+gst-build: replaced by mono repository
-- this section will be filled in in due course
-
-- FIXME: describe + link to Monorepo FAQ
+See mono repository section above and the GStreamer mono repository FAQ.
Cerbero
on platforms where dependencies are not readily available, such as
Windows, Android, iOS and macOS.
-General improvements
+General Cerbero improvements
-- this section will be filled in in due course
+- Plugin removed: libvisual
+- New plugins: rtpmanagerbad and rist
-macOS / iOS
+macOS / iOS specific Cerbero improvements
-- this section will be filled in in due course
+- XCode 12 support
+- macOS OS release support is now future-proof, similar to iOS
+- macOS Apple Silicon (ARM64) cross-compile support has been added
+- macOS Apple Silicon (ARM64) native support is currently experimental
-Windows
+Windows specific Cerbero improvements
-- this section will be filled in in due course
+- Visual Studio 2022 support has been added
+- bootstrap is faster since it requires building fewer build-tools
+ recipes on Windows
+- package is faster due to better scheduling of recipe stages and
+ elimination of unnecessary autotools regeneration
+- The following plugins are no longer built on Windows:
+ - a52dec (another decoder is still available in libav)
+ - dvdread
+ - resindvd
Windows MSI installer
-- this section will be filled in in due course
+- no major changes
-Linux
+Linux specific Cerbero improvements
-- this section will be filled in in due course
+- Fedora, Debian OS release support is now more future-proof
+- Amazon Linux 2 support has been added
-Android
+Android specific Cerbero improvements
-- this section will be filled in in due course
+- no major changes
Platform-specific changes and improvements
Android
-- this section will be filled in in due course
+- No major changes
macOS and iOS
-- this section will be filled in in due course
+- applemedia: add ProRes support to vtenc and vtdec
Windows
-- this section will be filled in in due course
+- On Windows the high-resolution clock is enabled now in the
+ gst-launch-1.0 and gst-play-1.0 command line tools, which provides
+ better clock and timer performance on Windows, at the cost of higher
+ power consumption. By default, without the high-resolution clock
+ enabled, the timer precision on Windows is system-dependent and may
+ be as bad as 15ms which is not good enough for many multimedia
+ applications. Developers may want to do the same in their Windows
+ applications if they think it’s a good idea for their application
+ use case, and depending on the Windows version they target. This is
+ not done automatically by GStreamer because on older Windows
+ versions (pre-Windows 10) this affects a global Windows setting and
+ also there’s a power consumption vs. performance trade-off that may
+ differ from application to application.
+
+- dxgiscreencapsrc now supports resolution changes
+
+- The wasapi2 audio plugin was rewritten and now has a higher rank
+ than the old wasapi plugin since it has a number of additional
+ features such as automatic stream routing, and no
+ known-but-hard-to-fix issues. The plugin is always built if the
+ Windows 10 SDK is available now.
+
+- The wasapi device providers now detect and notify dynamic device
+ additions/removals
+
+- d3d11screencapturesrc: new desktop capture element, including
+ GstDeviceProvider implementation to enumerate/select target monitors
+ for capture.
+
+- Direct3D11/DXVA decoder now supports AV1 and MPEG2 codecs
+ (d3d11av1dec, d3d11mpeg2dec)
+
+- VP9 decoding got more reliable and stable thanks to a newly written
+ codec parser
+
+- Support for decoding interlaced H.264/AVC streams
+
+- Hardware-accelerated video deinterlacing (d3d11deinterlace) and
+ video mixing (d3d11compositor)
+
+- Video mixing with the Direct3D11 API (d3d11compositor)
+
+- MediaFoundation API based hardware encoders gained the ability to
+ receive Direct3D11 textures as an input
+
+- Seungha’s blog post “GStreamer ❤ Windows: A primer on the cool stuff
+ you’ll find in the 1.20 release” describes many of the
+ Windows-related improvements in more detail
Linux
-- this section will be filled in in due course
+- bluez: LDAC Bluetooth audio codec support in a2dpsink and avdtpsink,
+ as well as an LDAC RTP payloader (rtpldacpay) and an LDAC audio
+ encoder (ldacenc)
+
+- kmssink: gained support for NV24, NV61, RGB16/BGR16 formats;
+ auto-detect NVIDIA Tegra driver
Documentation improvements
-- this section will be filled in in due course
+- hardware-accelerated GPU plugins will now no longer always list all
+ the element variants for all available GPUs, since those are
+ system-dependent and it’s confusing for users to see those in the
+ documentation just because the GStreamer developer who generated the
+ docs had multiple GPUs to play with at the time. Instead just show
+ the default elements.
-Possibly Breaking Changes
+Possibly Breaking and Other Noteworthy Behavioural Changes
+
+- gst_parse_launch(), gst_parse_bin_from_description() and friends
+ will now error out when setting properties that don’t exist on
+ top-level bins. They were silently ignored before.
+
+- The GstWebRTC library does not expose any objects anymore with
+ public fields. Instead properties have been added to replace that
+ functionality. If you are accessing such fields in your application,
+ switch to the corresponding properties.
-- FIXME: this section will be filled in in due course
-- MPEG-TS SCTE-35 API changes (FIXME: flesh out)
-- gst_parse_launch() and friends now error out on non-existing
- properties on top-level bins where they would silently fail and
- ignore those before.
- playbin and uridecodebin now emit the source-setup signal before the
element is added to the bin and linked so that the source element is
already configured before any scheduling query comes in, which is
- useful for elements such as appsrc or giostreamsrc. (Merge Request)
+ useful for elements such as appsrc or giostreamsrc.
+
+- The source element inside urisourcebin (used inside uridecodebin3
+ which is used inside playbin3) is no longer called "source". This
+ shouldn’t affect anyone hopefully, because there’s a "setup-source"
+ signal to configure the source element and no one should rely on
+ names of internal elements anyway.
+
+- The vp8enc element now expects bps (bits per second) for the
+ "temporal-scalability-target-bitrate" property, which is consistent
+ with the "target-bitrate" property. Since additional configuration
+ is required with modern libvpx to make temporal scaling work anyway,
+ chances are that very few people will have been using this property
+
+- vp8enc and vp9enc now default to “good quality” for the "deadline"
+ property rather then “best quality”. Having the deadline set to best
+ quality causes the encoder to be absurdly slow, most real-life users
+ will want the good quality tradeoff instead.
+
+- The experimental GstTranscoder library API in gst-plugins-bad was
+ changed from a GObject signal-based notification mechanism to a
+ GstBus/message-based mechanism akin to GstPlayer/GstPlay.
+
+- MPEG-TS SCTE-35 API: semantic change for SCTE-35 splice commands:
+ timestamps passed by the application should be in running time now,
+ since users of the API can’t really be expected to predict the local
+ PTS of the muxer.
+
+- The GstContext used by souphttpsrc to share the session between
+ multiple element instances has changed. Previously it provided
+ direct access to the internal SoupSession object, now it only
+ provides access to an opaque, internal type. This change is
+ necessary because SoupSession is not thread-safe at all and can’t be
+ shared safely between arbitrary external code and souphttpsrc.
+
+- Python bindings: GObject-introspection related Annotation fixes have
+ led to a case of a GstVideo.VideoInfo-related function signature
+ changing in the Python bindings (possibly one or two other cases
+ too). This is for a function that should never have been exposed in
+ the first place though, so the bindings are being updated to throw
+ an exception in that case, and the correct replacement API has been
+ added in form of an override.
Known Issues
-- this section will be filled in in due course
-
-- There are a couple of known WebRTC-related regressions/blockers:
-
- - webrtc: DTLS setup with Chrome is broken
- - webrtcbin: First keyframe is usually lost
+- nothing in particular at this point (but also see possibly breaking
+ changes section above)
Contributors
-- this section will be filled in in due course
+Aaron Boxer, Adam Leppky, Adam Williamson, Alba Mendez, Alejandro
+González, Aleksandr Slobodeniuk, Alexander Vandenbulcke, Alex Ashley,
+Alicia Boya García, Andika Triwidada, Andoni Morales Alastruey, Andrew
+Wesie, Andrey Moiseev, Antonio Ospite, Antonio Rojas, Arthur Crippa
+Búrigo, Arun Raghavan, Ashley Brighthope, Axel Kellermann, Baek, Bastien
+Nocera, Bastien Reboulet, Benjamin Gaignard, Bing Song, Binh Truong,
+Biswapriyo Nath, Brad Hards, Brad Smith, Brady J. Garvin, Branko
+Subasic, Camilo Celis Guzman, Chris Bass, ChrisDuncanAnyvision, Chris
+White, Corentin Damman, Daniel Almeida, Daniel Knobe, Daniel Stone,
+david, David Fernandez, David Keijser, David Phung, Devarsh Thakkar,
+Dinesh Manajipet, Dmitry Samoylov, Dmitry Shusharin, Dominique Martinet,
+Doug Nazar, Ederson de Souza, Edward Hervey, Emmanuel Gil Peyrot,
+Enrique Ocaña González, Ezequiel Garcia, Fabian Orccon, Fabrice
+Fontaine, Fernando Jimenez Moreno, Florian Karydes, Francisco Javier
+Velázquez-García, François Laignel, Frederich Munch, Fredrik Pålsson,
+George Kiagiadakis, Georg Lippitsch, Göran Jönsson, Guido Günther,
+Guillaume Desmottes, Guiqin Zou, Haakon Sporsheim, Haelwenn (lanodan)
+Monnier, Haihao Xiang, Haihua Hu, Havard Graff, He Junyan, Helmut
+Januschka, Henry Wilkes, Hosang Lee, Hou Qi, Ignacio Casal Quinteiro,
+Igor Kovalenko, Ilya Kreymer, Imanol Fernandez, Jacek Tomaszewski, Jade
+Macho, Jakub Adam, Jakub Janků, Jan Alexander Steffens (heftig), Jan
+Schmidt, Jason Carrete, Jason Pereira, Jay Douglass, Jeongki Kim, Jérôme
+Laheurte, Jimmi Holst Christensen, Johan Sternerup, John Hassell, John
+Lindgren, John-Mark Bell, Jonathan Matthew, Jordan Petridis, Jose
+Quaresma, Julian Bouzas, Julien, Kai Uwe Broulik, Kasper Steensig
+Jensen, Kellermann Axel, Kevin Song, Khem Raj, Knut Inge Hvidsten, Knut
+Saastad, Kristofer Björkström, Lars Lundqvist, Lawrence Troup, Lim Siew
+Hoon, Lucas Stach, Ludvig Rappe, Luis Paulo Fernandes de Barros, Luke
+Yelavich, Mads Buvik Sandvei, Marc Leeman, Marco Felsch, Marek Vasut,
+Marian Cichy, Marijn Suijten, Marius Vlad, Markus Ebner, Mart Raudsepp,
+Matej Knopp, Mathieu Duponchelle, Matthew Waters, Matthieu De Beule,
+Mengkejiergeli Ba, Michael de Gans, Michael Olbrich, Michael Tretter,
+Michal Dzik, Miguel Paris, Mikhail Fludkov, mkba, Nazar Mokrynskyi,
+Nicholas Jackson, Nicola Murino, Nicolas Dufresne, Niklas Hambüchen,
+Nikolay Sivov, Nirbheek Chauhan, Olivier Blin, Olivier Crete, Olivier
+Crête, Paul Goulpié, Per Förlin, Peter Boba, P H, Philippe Normand,
+Philipp Zabel, Pieter Willem Jordaan, Piotrek Brzeziński, Rafał
+Dzięgiel, Rafostar, raghavendra, Raghavendra, Raju Babannavar, Raleigh
+Littles III, Randy Li, Randy Li (ayaka), Ratchanan Srirattanamet, Raul
+Tambre, reed.lawrence, Ricky Tang, Robert Rosengren, Robert Swain, Robin
+Burchell, Roman Sivriver, R S Nikhil Krishna, Ruben Gonzalez, Ruslan
+Khamidullin, Sanchayan Maity, Scott Moreau, Sebastian Dröge, Sergei
+Kovalev, Seungha Yang, Sid Sethupathi, sohwan.park, Sonny Piers, Staz M,
+Stefan Brüns, Stéphane Cerveau, Stephan Hesse, Stian Selnes, Stirling
+Westrup, Théo MAILLART, Thibault Saunier, Tim, Timo Wischer, Tim-Philipp
+Müller, Tim Schneider, Tobias Ronge, Tom Schoonjans, Tulio Beloqui,
+tyler-aicradle, U. Artie Eoff, Ung, Val Doroshchuk, VaL Doroshchuk,
+Víctor Manuel Jáquez Leal, Vivek R, Vivia Nikolaidou, Vivienne
+Watermeier, Vladimir Menshakov, Will Miller, Wim Taymans, Xabier
+Rodriguez Calvar, Xavier Claessens, Xℹ Ruoyao, Yacine Bandou, Yinhang
+Liu, youngh.lee, youngsoo.lee, yychao, Zebediah Figura, Zhang yuankun,
+Zhang Yuankun, Zhao, Zhao Zhili, , Aleksandar Topic, Antonio Ospite,
+Bastien Nocera, Benjamin Gaignard, Brad Hards, Carlos Falgueras García,
+Célestin Marot, Corentin Damman, Corentin Noël, Daniel Almeida, Daniel
+Knobe, Danny Smith, Dave Piché, Dmitry Osipenko, Fabrice Fontaine,
+fjmax, Florian Zwoch, Guillaume Desmottes, Haihua Hu, Heinrich Kruger,
+He Junyan, Jakub Adam, James Cowgill, Jan Alexander Steffens (heftig),
+Jean Felder, Jeongki Kim, Jiri Uncovsky, Joe Todd, Jordan Petridis,
+Krystian Wojtas, Marc-André Lureau, Marcin Kolny, Marc Leeman, Mark
+Nauwelaerts, Martin Reboredo, Mathieu Duponchelle, Matthew Waters,
+Mengkejiergeli Ba, Michael Gruner, Nicolas Dufresne, Nirbheek Chauhan,
+Olivier Crête, Philippe Normand, Rafał Dzięgiel, Ralf Sippl, Robert
+Mader, Sanchayan Maity, Sangchul Lee, Sebastian Dröge, Seungha Yang,
+Stéphane Cerveau, Teh Yule Kim, Thibault Saunier, Thomas Klausner, Timo
+Wischer, Tim-Philipp Müller, Tobias Reineke, Tomasz Andrzejak, Trung Do,
+Tyler Compton, Ung, Víctor Manuel Jáquez Leal, Vivia Nikolaidou, Wim
+Taymans, wngecn, Wonchul Lee, wuchang li, Xavier Claessens, Xi Ruoyao,
+Yoshiharu Hirose, Zhao,
… and many others who have contributed bug reports, translations, sent
suggestions or helped testing.
1.20.0
-1.20.0 is scheduled to be released around October/November 2021.
+1.20.0 is scheduled to be released around early February 2022.
Schedule for 1.22
unstable development version leading up to the stable 1.22 release. The
development of 1.21/1.22 will happen in the git main branch.
-The plan for the 1.22 development cycle is yet to be confirmed.
+The plan for the 1.22 development cycle is yet to be confirmed. Assuming
+no major project-wide reorganisations in the 1.22 cycle we might try and
+aim for a release around August 2022.
1.22 will be backwards-compatible to the stable 1.20, 1.18, 1.16, 1.14,
1.12, 1.10, 1.8, 1.6, 1.4, 1.2 and 1.0 release series.
------------------------------------------------------------------------
These release notes have been prepared by Tim-Philipp Müller with
-contributions from …
+contributions from Matthew Waters, Nicolas Dufresne, Nirbheek Chauhan,
+Sebastian Dröge and Seungha Yang.
License: CC BY-SA 4.0
-This is GStreamer gst-plugins-ugly 1.19.3.
+This is GStreamer gst-plugins-ugly 1.19.90.
GStreamer 1.19 is the development branch leading up to the next major
stable version which will be 1.20.
<release>
<Version>
+ <revision>1.19.90</revision>
+ <branch>main</branch>
+ <name></name>
+ <created>2022-01-28</created>
+ <file-release rdf:resource="https://gstreamer.freedesktop.org/src/gst-plugins-ugly/gst-plugins-ugly-1.19.90.tar.xz" />
+ </Version>
+ </release>
+
+ <release>
+ <Version>
<revision>1.19.3</revision>
<branch>main</branch>
<name></name>
project('gst-plugins-ugly', 'c',
- version : '1.19.3.1',
+ version : '1.19.90',
meson_version : '>= 0.59',
default_options : [ 'warning_level=1',
'buildtype=debugoptimized' ])
+=== release 1.19.90 ===
+
+2022-01-28 14:28:35 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * NEWS:
+ * RELEASE:
+ * gst-python.doap:
+ * meson.build:
+ Release 1.19.90
+
+2022-01-28 14:28:28 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * ChangeLog:
+ Update ChangeLogs for 1.19.90
+
2022-01-26 00:02:49 +0100 Mathieu Duponchelle <mathieu@centricular.com>
* gi/overrides/GstAudio.py:
GStreamer 1.20 Release Notes
GStreamer 1.20 has not been released yet. It is scheduled for release in
-November 2021.
+late January / early February 2022.
1.19.x is the unstable development version that is being developed in
-the git main branch and which will eventually result in 1.20, and 1.19.3
-is the current development release in that series
-
-It is expected that feature freeze will be in early October 2021,
-followed by one or two 1.19.9x pre-releases and the new 1.20 stable
-release around October/November 2021.
+the git main branch and which will eventually result in 1.20, and
+1.19.90 is the first release candidate in that series (1.20rc1).
1.20 will be backwards-compatible to the stable 1.18, 1.16, 1.14, 1.12,
-1.10, 1.8, 1.6,, 1.4, 1.2 and 1.0 release series.
+1.10, 1.8, 1.6, 1.4, 1.2 and 1.0 release series.
See https://gstreamer.freedesktop.org/releases/1.20/ for the latest
version of this document.
-Last updated: Monday 1 November 2021, 01:00 UTC (log)
+Last updated: Wednesday 26 January 2022, 01:00 UTC (log)
Introduction
Highlights
-- this section will be completed in due course
+- Development in GitLab was switched to a single git repository
+ containing all the modules
+- GstPlay: new high-level playback library, replaces GstPlayer
+- WebM Alpha decoding support
+- Encoding profiles can now be tweaked with additional
+ application-specified element properties
+- Compositor: multi-threaded video conversion and mixing
+- RTP header extensions: unified support in RTP depayloader and
+ payloader base classes
+- SMPTE 2022-1 2-D Forward Error Correction support
+- Smart encoding (passthrough) support for VP8, VP9, H.265 in
+ encodebin and transcodebin
+- Runtime compatibility support for libsoup2 and libsoup3 (libsoup3
+ support experimental)
+- Video decoder subframe support
+- Video decoder automatic packet-loss, data corruption, and keyframe
+ request handling for RTP / WebRTC / RTSP
+- MP4 and Matroska muxers now support profile/level/resolution changes
+ for H264/H265 input streams (i.e. codec data changing on the fly)
+- MP4 muxing mode that initially creates a fragmented mp4 which is
+ converted to a regular mp4 on EOS
+- Audio support for the WebKit Port for Embedded (WPE) web page source
+ element
+- CUDA based video color space convert and rescale elements and
+ upload/download elements
+- NVIDIA memory:NVMM support for OpenGL glupload and gldownload
+ elements
+- Many WebRTC improvements
+- The new VA-API plugin implemention fleshed out with more decoders
+ and new postproc elements
+- AppSink API to retrieve events in addition to buffers and buffer
+ lists
+- AppSrc gained more configuration options for the internal queue
+ (leakiness, limits in buffers and time, getters to read current
+ levels)
+- Updated Rust bindings and many new Rust plugins
+- Improved support for custom minimal GStreamer builds
+- Support build against FFmpeg 5.0
+- Linux Stateless CODEC support gained MPEG2 and VP9
+- Windows Direct3D11/DXVA decoder gained AV1 and MPEG2 support
+- Lots of new plugins, features, performance improvements and bug
+ fixes
Major new features and changes
Noteworthy new features and API
-- this section will be filled in in due course
+- gst_element_get_request_pad() has been deprecated in favour of the
+ newly-added gst_element_request_pad_simple() which does the exact
+ same thing but has a less confusing name that hopefully makes clear
+ that the function request a new pad rather than just retrieves an
+ already-existing request pad.
+
+Development in GitLab was switched to a single git repository containing all the modules
+
+The GStreamer multimedia framework is a set of libraries and plugins
+split into a number of distinct modules which are released independently
+and which have so far been developed in separate git repositories in
+freedesktop.org GitLab.
+
+In addition to these separate git repositories there was a gst-build
+module that would use the Meson build systems’s subproject feature to
+download each individual module and then build everything in one go. It
+would also provide an uninstalled development environment that made it
+easy to work on GStreamer and use or test versions other than the
+system-installed GStreamer version.
+
+All of these modules have now (as of 28 September 2021) been merged into
+a single git repository (“Mono repository” or “monorepo”) which should
+simplify development workflows and continuous integration, especially
+where changes need to be made to multiple modules at once.
+
+This mono repository merge will primarily affect GStreamer developers
+and contributors and anyone who has workflows based on the GStreamer git
+repositories.
+
+The Rust bindings and Rust plugins modules have not been merged into the
+mono repository at this time because they follow a different release
+cycle.
+
+The mono repository lives in the existing GStreamer core git repository
+in GitLab in the new main branch and all future development will happen
+on this branch.
+
+Modules will continue to be released as separate tarballs.
+
+For more details, please see the GStreamer mono repository FAQ.
+
+GstPlay: new high-level playback library replacing GstPlayer
+
+- GstPlay is a new high-level playback library that replaces the older
+ GstPlayer API. It is basically the same API as GstPlayer but
+ refactored to use bus messages for application notifications instead
+ of GObject signals. There is still a signal adapter object for those
+ who prefer signals. Since the existing GstPlayer API is already in
+ use in various applications, it didn’t seem like a good idea to
+ break it entirely. Instead a new API was added, and it is expected
+ that this new GstPlay API will be moved to gst-plugins-base in
+ future.
+
+- The existing GstPlayer API is scheduled for deprecation and will be
+ removed at some point in the future (e.g. in GStreamer 1.24), so
+ application developers are urged to migrate to the new GstPlay API
+ at their earliest convenience.
+
+WebM alpha decoding
+
+- Implement WebM alpha decoding (VP8/VP9 with alpha), which required
+ support and additions in various places. This is supported both with
+ software decoders and hardware-accelerated decoders.
+
+- VP8/VP9 don’t support alpha components natively in the codec, so the
+ way this is implemented in WebM is by encoding the alpha plane with
+ transparency data as a separate VP8/VP9 stream. Inside the WebM
+ container (a variant of Matroska) this is coded as a single video
+ track with the “normal” VP8/VP9 video data making up the main video
+ data and each frame of video having an encoded alpha frame attached
+ to it as extra data ("BlockAdditional").
+
+- matroskademux has been extended extract this per-frame alpha side
+ data and attach it in form of a GstVideoCodecAlphaMeta to the
+ regular video buffers. Note that this new meta is specific to this
+ VP8/VP9 alpha support and can’t be used to just add alpha support to
+ other codecs that don’t support it. Lastly, matroskademux also
+ advertises the fact that the streams contain alpha in the caps.
+
+- The new codecalpha plugin contains various bits of infrastructure to
+ support autoplugging and debugging:
+
+ - codecalphademux splits out the alpha stream from the metas on
+ the regular VP8/VP9 buffers
+ - alphacombine takes two decoded raw video streams (one alpha, one
+ the regular video) and combines it into a video stream with
+ alpha
+ - vp8alphadecodebin + vp9alphadecodebin are wrapper bins that use
+ the regular vp8dec and vp9dec software decoders to decode
+ regular and alpha streams and combine them again. To decodebin
+ these look like regular decoders which ju
+ - The V4L2 CODEC plugin has stateless VP8/VP9 decoders that can
+ decode both alpha and non-alpha stream with a single decoder
+ instance
+
+- A new AV12 video format was added which is basically NV12 with an
+ alpha plane, which is more convenient for many hardware-accelerated
+ decoders.
+
+- Watch Nicolas Dufresne’s LCA 2022 talk “Bringing WebM Alpha support
+ to GStreamer” for all the details and a demo.
+
+RTP Header Extensions Base Class and Automatic Header Extension Handling in RTP Payloaders and Depayloaders
+
+- RTP Header Extensions are specified in RFC 5285 and provide a way to
+ add small pieces of data to RTP packets in between the RTP header
+ and the RTP payload. This is often used for per-frame metadata,
+ extended timestamps or other application-specific extra data. There
+ are several commonly-used extensions specified in various RFCs, but
+ senders are free to put any kind of data in there, as long as sender
+ and receiver both know what that data is. Receivers that don’t know
+ about the header extensions will just skip the extra data without
+ ever looking at it. These header extensions can often be combined
+ with any kind of payload format, so may need to be supported by many
+ RTP payloader and depayloader elements.
+
+- Inserting and extracting RTP header extension data has so far been a
+ bit inconvenient in GStreamer: There are functions to add and
+ retrieve RTP header extension data from RTP packets, but nothing
+ works automatically, even for common extensions. People would have
+ to do the insertion/extraction either in custom elements
+ before/after the RTP payloader/depayloader, or inside pad probes,
+ which isn’t very nice.
+
+- This release adds various pieces of new infrastructure for generic
+ RTP header extension handling, as well as some implementations for
+ common extensions:
+
+ - GstRTPHeaderExtension is a new helper base class for reading and
+ writing RTP header extensions. Nominally this subclasses
+ GstElement, but only so these extensions are stored in the
+ registry where they can be looked up by URI or name. They don’t
+ have pads and don’t get added to the pipeline graph as an
+ element.
+
+ - "add-extension" and "clear-extension" action signals on RTP
+ payloaders and depayloaders for manual extension management
+
+ - The "request-extension" signal will be emitted if an extension
+ is encountered that requires explicit mapping by the application
+
+ - new "auto-header-extension" property on RTP payloaders and
+ depayloaders for automatic handling of known header extensions.
+ This is enabled by default. The extensions must be signalled via
+ caps / SDP.
+
+ - RTP header extension implementations:
+
+ - rtphdrextclientaudiolevel: Client-to-Mixer Audio Level
+ Indication (RFC 6464) (also see below)
+ - rtphdrextcolorspace: Color Space extension, extends RTP
+ packets with color space and high dynamic range (HDR)
+ information
+ - rtphdrexttwcc: Transport Wide Congestion Control support
+
+- gst_rtp_buffer_remove_extension_data() is a new helper function to
+ remove an RTP header extension from an RTP buffer
+
+- The existing gst_rtp_buffer_set_extension_data() now also supports
+ shrinking the extension data in size
+
+AppSink and AppSrc improvements
+
+- appsink: new API to pull events out of appsink in addition to
+ buffers and buffer lists.
+
+ There was previously no way for users to receive incoming events
+ from appsink properly serialised with the data flow, even if they
+ are serialised events. The reason for that is that the only way to
+ intercept events was via a pad probe on the appsink sink pad, but
+ there is also internal queuing inside of appsink, so it’s difficult
+ to ascertain the right order of everything in all cases.
+
+ There is now a new "new-serialized-event" signal which will be
+ emitted when there’s a new event pending (just like the existing
+ "new-sample" signal). The "emit-signals" property must be set to
+ TRUE in order to activate this (but it’s also fine to just pull from
+ the application thread without using the signals).
+
+ gst_app_sink_pull_object() and gst_app_sink_try_pull_object() can be
+ used to pull out either an event or a new sample carrying a buffer
+ or buffer list, whatever is next in the queue.
+
+ EOS events will be filtered and will not be returned. EOS handling
+ can be done the ususal way, same as with _pull_sample().
+
+- appsrc: allow configuration of internal queue limits in time and
+ buffers and add leaky mode.
+
+ There is internal queuing inside appsrc so the application thread
+ can push data into the element which will then be picked up by the
+ source element’s streaming thread and pushed into the pipeline from
+ that streaming thread. This queue is unlimited by default and until
+ now it was only possible to set a maximum size limit in bytes. When
+ that byte limit is reached, the pushing thread (application thread)
+ would be blocked until more space becomes available.
+
+ A limit in bytes is not particularly useful for many use cases, so
+ now it is possible to also configure limits in time and buffers
+ using the new "max-time" and "max-buffers" properties. Of course
+ there are also matching new read-only"current-level-buffers" and
+ "current-level-time properties" properties to query the current fill
+ level of the internal queue in time and buffers.
+
+ And as if that wasn’t enough the internal queue can also be
+ configured as leaky using the new "leaky-type" property. That way
+ when the queue is full the application thread won’t be blocked when
+ it tries to push in more data, but instead either the new buffer
+ will be dropped or the oldest data in the queue will be dropped.
+
+Better string serialization of nested GstCaps and GstStructures
+
+- New string serialisation format for structs and caps that can handle
+ nested structs and caps properly by using brackets to delimit nested
+ items (e.g. some-struct, some-field=[nested-struct, nested=true]).
+ Unlike the default format the new variant can also support more than
+ one level of nesting. For backwards-compatibility reasons the old
+ format is still output by default when serialising caps and structs
+ using the existing API. The new functions gst_caps_serialize() and
+ gst_structure_serialize() can be used to output strings in the new
+ format.
+
+Convenience API for custom GstMetas
+
+- New convenience API to register and create custom GstMetas:
+ gst_meta_register_custom() and gst_buffer_add_custom_meta(). Such
+ custom meta is backed by a GstStructure and does not require that
+ users of the API expose their GstMeta implementation as public API
+ for other components to make use of it. In addition, it provides a
+ simpler interface by ignoring the impl vs. api distinction that the
+ regular API exposes. This new API is meant to be the meta
+ counterpart to custom events and messages, and to be more convenient
+ than the lower-level API when the absolute best performance isn’t a
+ requirement. The reason it’s less performant than a “proper” meta is
+ that a proper meta is just a C struct in the end whereas this goes
+ through the GstStructure API which has a bit more overhead, which
+ for most scenarios is negligible however. This new API is useful for
+ experimentation or proprietary metas, but also has some limitations:
+ it can only be used if there’s a single producer of these metas;
+ it’s not allowed to register the same custom meta multiple times or
+ from multiple places.
+
+Additional Element Properties on Encoding Profiles
+
+- GstEncodingProfile: The new "element-properties" and
+ gst_encoding_profile_set_element_properties() API allows
+ applications to set additional element properties on encoding
+ profiles to configure muxers and encoders. So far the encoding
+ profile template was the only place where this could be specified,
+ but often what applications want to do is take a ready-made encoding
+ profile shipped by GStreamer or the application and then tweak the
+ settings on top of that, which is now possible with this API. Since
+ applications can’t always know in advance what encoder element will
+ be used in the end, it’s even possible to specify properties on a
+ per-element basis.
+
+ Encoding Profiles are used in the encodebin, transcodebin and
+ camerabin elements and APIs to configure output formats (containers
+ and elementary streams).
+
+Audio Level Indication Meta for RFC 6464
+
+- New GstAudioLevelMeta containing Audio Level Indication as per RFC
+ 6464
+
+- The level element has been updated to add GstAudioLevelMeta on
+ buffers if the "audio-level-meta" property is set to TRUE. This can
+ then in turn be picked up by RTP payloaders to signal the audio
+ level to receivers through RTP header extensions (see above).
+
+- New Client-to-Mixer Audio Level Indication (RFC6464) RTP Header
+ Extension which should be automatically created and used by RTP
+ payloaders and depayloaders if their "auto-header-extension"
+ property is enabled and if the extension is part of the RTP caps.
+
+Automatic packet loss, data corruption and keyframe request handling for video decoders
+
+- The GstVideoDecoder base class has gained various new APIs to
+ automatically handle packet loss and data corruption better by
+ default, especially in RTP, RTSP and WebRTC streaming scenarios, and
+ to give subclasses more control about how they want to handle
+ missing data:
+
+ - Video decoder subclasses can mark output frames as corrupted via
+ the new GST_VIDEO_CODEC_FRAME_FLAG_CORRUPTED flag
+
+ - A new "discard-corrupted-frames" property allows applications to
+ configure decoders so that corrupted frames are directly
+ discarded instead of being forwarded inside the pipeline. This
+ is a replacement for the "output-corrupt" property of the FFmpeg
+ decoders.
+
+ - RTP depayloaders can now signal to decoders that data is missing
+ when sending GAP events for lost packets. GAP events can be sent
+ for various reason in a GStreamer pipeline. Often they are just
+ used to let downstream elements know that there isn’t a buffer
+ available at the moment, so downstream elements can move on
+ instead of waiting for one. They are also sent by RTP
+ depayloaders in the case that packets are missing, however, and
+ so far a decoder was not able to differentiate the two cases.
+ This has been remedied now: GAP events can be decorated with
+ gst_event_set_gap_flags() and GST_GAP_FLAG_MISSING_DATA to let
+ decoders now what happened, and decoders can then use that in
+ some cases to handle missing data better.
+
+ - The GstVideoDecoder::handle_missing_data vfunc was added to
+ inform subclasses about packet loss or missing data and let them
+ handle it in their own way if they like.
+
+ - gst_video_decoder_set_needs_sync_point() lets subclasses signal
+ that they need the stream to start with a sync point. If
+ enabled, the base class will discard all non-sync point frames
+ in the beginning and after a flush and does not pass them to the
+ subclass. Furthermore, if the first frame is not a sync point,
+ the base class will try and request a sync frame from upstream
+ by sending a force-key-unit event (see next items).
+
+ - New "automatic-request-sync-points" and
+ "automatic-request-sync-point-flags" properties to automatically
+ request sync points when needed, e.g. on packet loss or if the
+ first frame is not a keyframe. Applications may want to enable
+ this on decoders operating in e.g. RTP/WebRTC/RTSP receiver
+ pipelines.
+
+ - The new "min-force-key-unit-interval" property can be used to
+ ensure there’s a minimal interval between keyframe requests to
+ upstream (and/or the sender) and we’re not flooding the sender
+ with key unit requests.
+
+ - gst_video_decoder_request_sync_point() allows subclasses to
+ request a new sync point (e.g. if they choose to do their own
+ missing data handling). This will still honour the
+ "min-force-key-unit-interval" property if set.
+
+Improved support for custom minimal GStreamer builds
+
+- Element registration and registration of other plugin features
+ inside plugin init functions has been improved in order to
+ facilitate minimal custom GStreamer builds.
+
+- A number of new macros have been added to declare and create
+ per-element and per-pluginfeature register functions in all plugins,
+ and then call those from the per-plugin plugin_init functions:
+
+ - GST_ELEMENT_REGISTER_DEFINE,
+ GST_DEVICE_PROVIDER_REGISTER_DEFINE,
+ GST_DYNAMIC_TYPE_REGISTER_DEFINE, GST_TYPE_FIND_REGISTER_DEFINE
+ for the actual registration call with GStreamer
+ - GST_ELEMENT_REGISTER, GST_DEVICE_PROVIDER_REGISTER,
+ GST_DYNAMIC_TYPE_REGISTER, GST_PLUGIN_STATIC_REGISTER,
+ GST_TYPE_FIND_REGISTER to call the registration function defined
+ by the REGISTER_DEFINE macro
+ - GST_ELEMENT_REGISTER_DECLARE,
+ GST_DEVICE_PROVIDER_REGISTER_DECLARE,
+ GST_DYNAMIC_TYPE_REGISTER_DECLARE,
+ GST_TYPE_FIND_REGISTER_DECLARE to declare the registration
+ function defined by the REGISTER_DEFINE macro
+ - and various variants for advanced use cases.
+
+- This means that applications can call the per-element and
+ per-pluginfeature registration functions for only the elements they
+ need instead of registering plugins as a whole with all kinds of
+ elements that may not be required (e.g. encoder and decoder instead
+ of just decoder). In case of static linking all unused functions and
+ their dependencies would be removed in this case by the linker,
+ which helps minimise binary size for custom builds.
+
+- gst_init() will automatically call a gst_init_static_plugins()
+ function if one exists.
+
+- See the GStreamer static build documentation and Stéphane’s blog
+ post Generate a minimal GStreamer build, tailored to your needs for
+ more details.
New elements
-- this section will be filled in in due course
+- New aesdec and aesenc elements for AES encryption and decryption in
+ a custom format.
+
+- New encodebin2 element with dynamic/sometimes source pads in order
+ to support the option of doing the muxing outside of encodebin,
+ e.g. in combination with a splitmuxsink.
+
+- New fakeaudiosink and videocodectestsink elements for testing and
+ debugging (see below for more details)
+
+- rtpisacpay, rtpisacdepay: new RTP payloader and depayloader for iSAC
+ audio codec
+
+- rtpst2022-1-fecdec, rtpst2022-1-fecenc: new elements providing SMPTE
+ 2022-1 2-D Forward Error Correction. More details in Mathieu’s blog
+ post.
+
+- isac: new plugin wrapping the Internet Speech Audio Codec reference
+ encoder and decoder from the WebRTC project.
+
+- asio: plugin for Steinberg ASIO (Audio Streaming Input/Output) API
+
+- gssrc, gssink: add source and sink for Google Cloud Storage
+
+- onnx: new plugin to apply ONNX neural network models to video
+
+- openaptx: aptX and aptX-HD codecs using libopenaptx (v0.2.0)
+
+- qroverlay, debugqroverlay: new elements that allows overlaying data
+ on top of video in form of a QR code
+
+- cvtracker: new OpenCV-based tracker element
+
+- av1parse, vp9parse: new parsers for AV1 and VP9 video
+
+- va: work on the new VA-API plugin implementation for
+ hardware-accelerated video decoding and encoding has continued at
+ pace, with various new decoders and filters having joined the
+ initial vah264dec:
+
+ - vah265dec: VA-API H.265 decoder
+ - vavp8dec: VA-API VP8 decoder
+ - vavp9dec: VA-API VP9 decoder
+ - vaav1dec: VA-API AV1 decoder
+ - vampeg2dec: VA-API MPEG-2 decoder
+ - vadeinterlace: : VA-API deinterlace filter
+ - vapostproc: : VA-API postproc filter (color conversion,
+ resizing, cropping, color balance, video rotation, skin tone
+ enhancement, denoise, sharpen)
+
+ See Víctor’s blog post “GstVA in GStreamer 1.20” for more details
+ and what’s coming up next.
+
+- vaapiav1dec: new AV1 decoder element (in gstreamer-vaapi)
+
+- msdkav1dec: hardware-accelerated AV1 decoder using the Intel Media
+ SDK / oneVPL
+
+- nvcodec plugin for NVIDIA NVCODEC API for hardware-accelerated video
+ encoding and decoding:
+
+ - cudaconvert, cudascale: new CUDA based video color space convert
+ and rescale elements
+ - cudaupload, cudadownload: new helper elements for memory
+ transfer between CUDA and system memory spaces
+ - nvvp8sldec, nvvp9sldec: new GstCodecs-based VP8/VP9 decoders
+
+- Various new hardware-accelerated elements for Windows:
+
+ - d3d11screencapturesrc: new desktop capture element, including a
+ GstDeviceProvider implementation to enumerate/select target
+ monitors for capture.
+ - d3d11av1dec and d3d11mpeg2dec: AV1 and MPEG-2 decoders
+ - d3d11deinterlace: deinterlacing filter
+ - d3d11compositor: video composing element
+ - see Windows section below for more details
+
+- new Rust plugins:
+
+ - audiornnoise: Removes noise from an audio stream
+ - awstranscribeparse: Parses AWS audio transcripts into timed text
+ buffers
+ - ccdetect: Detects if valid closed captions are present in a
+ closed captions stream
+ - cea608tojson: Converts CEA-608 Closed Captions to a JSON
+ representation
+ - cmafmux: CMAF fragmented MP4 muxer
+ - dashmp4mux: DASH fragmented MP4 muxer
+ - isofmp4mux: ISO fragmented MP4 muxer
+ - ebur128level: EBU R128 Loudness Level Measurement
+ - ffv1dec: FFV1 video decoder
+ - gtk4paintablesink: GTK4 video sink, which provides a
+ GdkPaintable that can be rendered in various widgets
+ - hlssink3: HTTP Live Streaming sink
+ - hrtfrender: Head-Related Transfer Function (HRTF) renderer
+ - hsvdetector: HSV colorspace detector
+ - hsvfilter: HSV colorspace filter
+ - jsongstenc: Wraps buffers containing any valid top-level JSON
+ structures into higher level JSON objects, and outputs those as
+ ndjson
+ - jsongstparse: Parses ndjson as output by jsongstenc
+ - jsontovtt: converts JSON to WebVTT subtitles
+ - regex: Applies regular expression operations on text
+ - roundedcorners: Adds rounded corners to video
+ - spotifyaudiosrc: Spotify source
+ - textahead: Display upcoming text buffers ahead (e.g. for
+ Karaoke)
+ - transcriberbin: passthrough bin that transcribes raw audio to
+ closed captions using awstranscriber and puts the captions as
+ metas onto the video
+ - tttojson: Converts timed text to a JSON representation
+ - uriplaylistbin: Playlist source bin
+ - webpdec-rs: WebP image decoder with animation support
+
+- New plugin codecalpha with elements to assist with WebM Alpha
+ decoding
+
+ - codecalphademux: Split stream with GstVideoCodecAlphaMeta into
+ two streams
+ - alphacombine: Combine two raw video stream (I420 or NV12) as one
+ stream with alpha channel (A420 or AV12)
+ - vp8alphadecodebin: A bin to handle software decoding of VP8 with
+ alpha
+ - vp9alphadecodebin: A bin to handle software decoding of VP9 with
+ alpha
+
+- New hardware accelerated elements for Linux:
+
+ - v4l2slmpeg2dec: Support for Linux Stateless MPEG2 decoders
+ - v4l2slvp9dec: Support for Linux Stateless VP9 decoders
+ - v4l2slvp8alphadecodebin: Support HW accelerated VP8 with alpha
+ layer decoding
+ - v4l2slvp9alphadecodebin: Support HW accelerated VP9 with alpha
+ layer decoding
New element features and additions
-- this section will be filled in in due course
+- assrender: handle more font mime types; better interaction with
+ matroskademux for embedded fonts
+
+- audiobuffersplit: Add support for specifying output buffer size in
+ bytes (not just duration)
+
+- audiolatency: new "samplesperbuffer" property so users can configure
+ the number of samples per buffer. The default value is 240 samples
+ which is equivalent to 5ms latency with a sample rate of 48000,
+ which might be larger than actual buffer size of audio capture
+ device.
+
+- audiomixer, audiointerleave, GstAudioAggregator: now keep a count of
+ samples that are dropped or processed as statistic and can be made
+ to post QoS messages on the bus whenever samples are dropped by
+ setting the "qos-messages" property on input pads.
+
+- audiomixer, compositor: improved handling of new inputs added at
+ runtime. New API was added to the GstAggregator base class to allow
+ subclasses to opt into an aggregation mode where inactive pads are
+ ignored when processing input buffers
+ (gst_aggregator_set_ignore_inactive_pads(),
+ gst_aggregator_pad_is_inactive()). An “inactive pad” in this context
+ is a pad which, in live mode, hasn’t yet received a first buffer,
+ but has been waited on at least once. What would happen usually in
+ this case is that the aggregator would wait for data on this pad
+ every time, up to the maximum configured latency. This would
+ inadvertently push mixer elements in live mode to the configured
+ latency envelope and delay processing when new inputs are added at
+ runtime until these inputs have actually produced data. This is
+ usually undesirable. With this new API, new inputs can be added
+ (requested) and configured and they won’t delay the data processing.
+ Applications can opt into this new behaviour by setting the
+ "ignore-inactive-pads" property on compositor, audiomixer or other
+ GstAudioAggregator-based elements.
+
+- cccombiner: implement “scheduling” of captions. So far cccombiner’s
+ behaviour was essentially that of a funnel: it strictly looked at
+ input timestamps to associate together video and caption buffers.
+ Now it will try to smoothly schedule caption buffers in order to
+ have exactly one per output video buffer. This might involve
+ rewriting input captions, for example when the input is CDP then
+ sequence counters are rewritten, time codes are dropped and
+ potentially re-injected if the input video frame had a time code
+ meta. This can also lead to the input drifting from synchronization,
+ when there isn’t enough padding in the input stream to catch up. In
+ that case the element will start dropping old caption buffers once
+ the number of buffers in its internal queue reaches a certain limit
+ (configurable via the "max-scheduled" property). The new original
+ funnel-like behaviour can be restored by setting the "scheduling"
+ property to FALSE.
+
+- ccconverter: new "cdp-mode" property to specify which sections to
+ include in CDP packets (timecode, CC data, service info). Various
+ software, including ffmpeg’s Decklink support, fails parsing CDP
+ packets that contain anything but CC data in the CDP packets.
+
+- clocksync: new "sync-to-first" property for automatic timestamp
+ offset setup: if set clocksync will set up the "ts-offset" value
+ based on the first buffer and the pipeline’s running time when the
+ first buffer arrived. The newly configured "ts-offset" in this case
+ would be the value that allows outputting the first buffer without
+ waiting on the clock. This is useful for example to feed a non-live
+ input into an already-running pipeline.
+
+- compositor:
+
+ - multi-threaded input conversion and compositing. Set the
+ "max-threads" property to activate this.
+ - new "sizing-policy" property to support display aspect ratio
+ (DAR)-aware scaling. By default the image is scaled to fill the
+ configured destination rectangle without padding and without
+ keeping the aspect ratio. With sizing-policy=keep-aspect-ratio
+ the input image is scaled to fit the destination rectangle
+ specified by GstCompositorPad:{xpos, ypos, width, height}
+ properties preserving the aspect ratio. As a result, the image
+ will be centered in the destination rectangle with padding if
+ necessary.
+ - new "zero-size-is-unscaled" property on input pads. By default
+ pad width=0 or pad height=0 mean that the stream should not be
+ scaled in that dimension. But if the "zero-size-is-unscaled"
+ property is set to FALSE a width or height of 0 is instead
+ interpreted to mean that the input image on that pad should not
+ be composited, which is useful when creating animations where an
+ input image is made smaller and smaller until it disappears.
+ - improved handling of new inputs at runtime via
+ "ignore-inactive-pads"property (see above for details)
+ - allow output format with alpha even if none of the inputs have
+ alpha (also glvideomixer and other GstVideoAggregator
+ subclasses)
+
+- dashsink: add h265 codec support and signals for allowing custom
+ playlist/fragment output
+
+- decodebin3:
+
+ - improved decoder selection, especially for hardware decoders
+ - make input activation “atomic” when adding inputs dynamically
+ - better interleave handling: take into account decoder latency
+ for interleave size
+
+- decklink:
+
+ - Updated DeckLink SDK to 11.2 to support DeckLink 8K Pro
+ - decklinkvideosrc:
+ - More accurate and stable capture timestamps: use the
+ hardware reference clock time when the frame was finished
+ being captured instead of a clock time much further down the
+ road.
+ - Automatically detect widescreen vs. normal NTSC/PAL
+
+- encodebin:
+
+ - add “smart encoding” support for H.265, VP8 and VP9 (i.e. only
+ re-encode where needed and otherwise pass through encoded video
+ as-is).
+ - H264/H265 smart encoding improvements: respect user-specified
+ stream-format, but if not specified default to avc3/hvc1 with
+ in-band SPS/PPS/VPS signalling for more flexibility.
+ - new encodebin2 element with dynamic/sometimes source pads in
+ order to support the option of doing the muxing outside of
+ encodebin, e.g. in combination with splitmuxsink.
+ - add APIs to set element properties on encoding profiles (see
+ below)
+
+- errorignore: new "ignore-eos" property to also ignore FLOW_EOS from
+ downstream elements
+
+- giosrc: add support for growing source files: applications can
+ specify that the underlying file being read is growing by setting
+ the "is-growing" property. If set, the source won’t EOS when it
+ reaches the end of the file, but will instead start monitoring it
+ and will start reading data again whenever a change is detected. The
+ new "waiting-data" and "done-waiting-data" signals keep the
+ application informed about the current state.
+
+- gtksink, gtkglsink:
+
+ - scroll event support: forwarded as navigation events into the
+ pipeline
+ - "video-aspect-ratio-override" property to force a specific
+ aspect ratio
+ - "rotate-method" property and support automatic rotation based on
+ image tags
+
+- identity: new "stats" property allows applications to retrieve the
+ number of bytes and buffers that have passed through so far.
+
+- interlace: add support for more formats, esp 10-bit, 12-bit and
+ 16-bit ones
+
+- jack: new "low-latency" property for automatic latency-optimized
+ setting and "port-names" property to select ports explicitly
+
+- jpegdec: support output conversion to RGB using libjpeg-turbo (for
+ certain input files)
+
+- line21dec:
+
+ - "mode" property to control whether and how detected closed
+ captions should be inserted in the list of existing close
+ caption metas on the input frame (if any): add, drop, or
+ replace.
+ - "ntsc-only" property to only look for captions if video has NTSC
+ resolution
+
+- line21enc: new "remove-caption-meta" to remove metas from output
+ buffers after encoding the captions into the video data; support for
+ CDP closed captions
+
+- matroskademux, matroskamux: Add support for ffv1, a lossless
+ intra-frame video coding format.
+
+- matroskamux: accept in-band SPS/PPS/VPS for H264 and H265
+ (i.e. stream-format avc3 and hev1) which allows on-the-fly
+ profile/level/resolution changes.
+
+- matroskamux: new "cluster-timestamp-offset" property, useful for use
+ cases where the container timestamps should map to some absolute
+ wall clock time, for example.
+
+- rtpsrc: add "caps" property to allow explicit setting of the caps
+ where needed
+
+- mpegts: support SCTE-35 passthrough via new "send-scte35-events"
+ property on MPEG-TS demuxer tsdemux. When enabled, SCTE 35 sections
+ (eg ad placement opportunities) are forwarded as events donwstream
+ where they can be picked up again by mpegtsmux. This required a
+ semantic change in the SCTE-35 section API: timestamps are now in
+ running time instead of muxer pts.
+
+- tsdemux: Handle PCR-less MPEG-TS streams; more robust timestamp
+ handling in certain corner cases and for poorly muxed streams.
+
+- mpegtsmux:
+
+ - More conformance improvements to make MPEG-TS analyzers happy:
+ - PCR timing accuracy: Improvements to the way mpegtsmux
+ outputs PCR observations in CBR mode, so that a PCR
+ observation is always inserted when needed, so that we never
+ miss the configured pcr-interval, as that triggers various
+ MPEG-TS analyser errors.
+ - Improved PCR/SI scheduling
+ - Don’t write PCR until PAT/PMT are output to make sure streams
+ start cleanly with a PAT/PMT.
+ - Allow overriding the automatic PMT PID selection via
+ application-supplied PMT_%d fields in the prog-map
+ structure/property.
+
+- mp4mux:
+
+ - new "first-moov-then-finalise" mode for fragmented output where
+ the output will start with a self-contained moov atom for the
+ first fragment, and then produce regular fragments. Then at the
+ end when the file is finalised, the initial moov is invalidated
+ and a new moov is written covering the entire file. This way the
+ file is a “fragmented mp4” file while it is still being written
+ out, and remains playable at all times, but at the end it is
+ turned into a regular mp4 file (with former fragment headers
+ remaining as unused junk data in the file).
+ - support H.264 avc3 and H.265 hvc1 stream formats as input where
+ the codec data is signalled in-band inside the bitstream instead
+ of caps/file headers.
+ - support profile/level/resolution changes for H264/H265 input
+ streams (i.e. codec data changing on the fly). Each codec_data
+ is put into its own SampleTableEntry inside the stsd, unless the
+ input is in avc3 stream format in which case it’s written
+ in-band an not in the headers.
+
+- multifilesink: new ""min-keyframe-distance"" property to make
+ minimum distance between keyframes in next-file=key-frame mode
+ configurable instead of hard-coding it to 10 seconds.
+
+- mxfdemux has seen a big refactoring to support non-frame wrappings
+ and more accurate timestamp/seek handling for some formats
+
+- msdk plugin for hardware-accelerated video encoding and decoding
+ using the Intel Media SDK:
+
+ - oneVPL support (Intel oneAPI Video Processing Library)
+ - AV1 decoding support
+ - H.264 decoder now supports constrained-high and progressive-high
+ profiles
+ - H.264 encoder:
+ - more configuration options (properties):
+ "intra-refresh-type", "min-qp" , "max-qp", "p-pyramid",
+ "dblk-idc"
+ - H.265 encoder:
+ - can output main-still-picture profile
+ - now inserts HDR SEIs (mastering display colour volume and
+ content light level)
+ - more configuration options (properties):
+ "intra-refresh-type", "min-qp" , "max-qp", "p-pyramid",
+ "b-pyramid", "dblk-idc", "transform-skip"
+ - support for RGB 10bit format
+ - External bitrate control in encoders
+ - Video post proc element msdkvpp gained support for 12-bit pixel
+ formats P012_LE, Y212_LE and Y412_LE
+
+- nvh264sldec: interlaced stream support
+
+- openh264enc: support main, high, constrained-high and
+ progressive-high profiles
+
+- openjpeg: support for multithreaded decoding and encoding
+
+- rtspsrc: now supports IPv6 also for tunneled mode (RTSP-over-HTTP);
+ new "ignore-x-server-reply" property to ignore the
+ x-server-ip-address server header reply in case of HTTP tunneling,
+ as it is often broken.
+
+- souphttpsrc: Runtime compatibility support for libsoup2 and
+ libsoup3. libsoup3 is the latest major version of libsoup, but
+ libsoup2 and libsoup3 can’t co-exist in the same process because
+ there is no namespacing or versioning for GObject types. As a
+ result, it would be awkward if the GStreamer souphttpsrc plugin
+ linked to a specific version of libsoup, because it would only work
+ with applications that use the same version of libsoup. To make this
+ work, the soup plugin now tries to determine the libsoup version
+ used by the application (and its other dependencies) at runtime on
+ systems where GStreamer is linked dynamically. libsoup3 support is
+ still considered somewhat experimental at this point.
+
+- srtsrc, srtsink: add signals for the application to accept/reject
+ incoming connections
+
+- timeoverlay: new elapsed-running-time time mode which shows the
+ running time since the first running time (and each flush-stop).
+
+- udpsrc: new timestamping mode to retrieve packet receive timestamps
+ from the kernel via socket control messages (SO_TIMESTAMPNS) on
+ supported platforms
+
+- uritranscodebin: new setup-source and element-setup signals for
+ applications to configure elements used
+
+- v4l2codecs plugin gained support for 4x4 and 32x32 tile formats
+ enabling some platforms or direct renders. Important memory usage
+ improvement.
+
+- v4l2slh264dec now implements the final Linux uAPI as shipped on
+ Linux 5.11 and later.
+
+- valve: add "drop-mode" property and provide two new modes of
+ operation: in drop-mode=forward-sticky-events sticky events
+ (stream-start, segment, tags, caps, etc.) are forwarded downstream
+ even when dropping is enabled; drop-mode=transform-to-gap will in
+ addition also convert buffers into gap events when dropping is
+ enabled, which lets downstream elements know that time is advancing
+ and might allow for preroll in many scenarios. By default all events
+ and all buffers are dropped when dropping is enabled, which can
+ cause problems with caps negotiation not progressing or branches not
+ prerolling when dropping is enabled.
+
+- videocrop: support for many more pixel formats, e.g. planar YUV
+ formats with > 8bits and GBR* video formats; can now also accept
+ video not backed by system memory as long as downstream supports the
+ GstCropMeta
+
+- videotestsrc: new smpte-rp-219 pattern for SMPTE75 RP-219 conformant
+ color bars
+
+- vp8enc: finish support for temporal scalability: two new properties
+ ("temporal-scalability-layer-flags",
+ "temporal-scalability-layer-sync-flags") and a unit change on the
+ "temporal-scalability-target-bitrate" property (now expects bps);
+ also make temporal scalability details available to RTP payloaders
+ as buffer metadata.
+
+- vp9enc: new properties to tweak encoder performance:
+
+ - "aq-mode" to configure adaptive quantization modes
+ - "frame-parallel-decoding" to configure whether to create a
+ bitstream that reduces decoding dependencies between frames
+ which allows staged parallel processing of more than one video
+ frames in the decoder. (Defaults to TRUE)
+ - "row-mt", "tile-columns" and "tile-rows" so multithreading can
+ be enabled on a per-tile basis, instead of on a per tile-column
+ basis. In combination with the new "tile-rows" property, this
+ allows the encoder to make much better use of the available CPU
+ power.
+
+- vp9dec, vp9enc: add support for 10-bit 4:2:0 and 4:2:2 YUV, as well
+ as 8-bit 4:4:4
+
+- vp8enc, vp9enc now default to “good quality” for the deadline
+ property rather then “best quality”. Having the deadline set to best
+ quality causes the encoder to be absurdly slow, most real-life users
+ will prefer good-enough quality with better performance instead.
+
+- wpesrc:
+
+ - implement audio support: a new sometimes source pad will be
+ created for each audio stream created by the web engine.
+ - move wpesrc to wpevideosrc and add a wrapper bin wpesrc to also
+ support audio
+ - also handles web:// URIs now (same as cefsrc)
+ - post messages with the estimated load progress on the bus
-Plugin and library moves
+- x265enc: add negative DTS support, which means timestamps are now
+ offset by 1h same as with x264enc
+
+RTP Payloaders and Depayloaders
+
+- rtpisacpay, rtpisacdepay: new RTP payloader and depayloader for iSAC
+ audio codec
+
+- rtph264depay:
+
+ - new "request-keyframe" property to make the depayloader
+ automatically request a new keyframe from the sender on packet
+ loss, consistent with the new property on rtpvp8depay.
+ - new "wait-for-keyframe" property to make depayloader wait for a
+ new keyframe at the beginning and after packet loss (only
+ effective if the depayloader outputs AUs), consistent with the
+ existing property on rtpvp8depay.
+
+- rtpopuspay, rtpopusdepay: support libwebrtc-compatible multichannel
+ audio in addition to the previously supported multichannel audio
+ modes
+
+- rtpopuspay: add DTX (Discontinuous Transmission) support
+
+- rtpvp8depay: new "request-keyframe" property to make the depayloader
+ automatically request a new keyframe from the sender on packet loss.
+
+- rtpvp8pay: temporal scaling support
-- this section will be filled in in due course
+- rtpvp9depay: Improved SVC handling (aggregate all layers)
+
+RTP Infrastructure
+
+- rtpst2022-1-fecdec, rtpst2022-1-fecenc: new elements providing SMPTE
+ 2022-1 2-D Forward Error Correction. More details in Mathieu’s blog
+ post.
+
+- rtpreddec: BUNDLE support
+
+- rtpredenc, rtpulpfecenc: add support for Transport-wide Congestion
+ Control (TWCC)
+
+- rtpsession: new "twcc-feedback-interval" property to allow RTCP TWCC
+ reports to be scheduled on a timer instead of per marker-bit.
+
+Plugin and library moves
- There were no plugin moves or library moves in this cycle.
The following elements or plugins have been removed:
-- this section will be filled in in due course
-
- The ofa audio fingerprinting plugin has been removed. The MusicIP
database has been defunct for years so this plugin is likely neither
useful nor used by anyone.
Miscellaneous API additions
-- this section will be filled in in due course
+Core
+
+- gst_buffer_new_memdup() is a convenience function for the
+ widely-used gst_buffer_new_wrapped(g_memdup(data,size),size)
+ pattern.
+
+- gst_caps_features_new_single() creates a new single GstCapsFeatures,
+ avoiding the need to use the vararg function with NULL terminator
+ for simple cases.
+
+- gst_element_type_set_skip_documentation() can be used by plugins to
+ signal that certain elements should not be included in the GStreamer
+ plugin documentation. This is useful for plugins where elements are
+ registered dynamically based on hardware capabilities and/or where
+ the available plugins and properties vary from system to system.
+ This is used in the d3d11 plugin for example to ensure that only the
+ list of default elements is advertised in the documentation.
+
+- gst_type_find_suggest_empty_simple() is a new convenience function
+ for typefinders for cases where there’s only a media type and no
+ other fields.
+
+- New API to create elements and set properties at construction time,
+ which is not only convenient, but also allows GStreamer elements to
+ have construct-only properties: gst_element_factory_make_full(),
+ gst_element_factory_make_valist(),
+ gst_element_factory_make_with_properties(),
+ gst_element_factory_create_full(),
+ gst_element_factory_create_valist(),
+ gst_element_factory_create_with_properties().
+
+- GstSharedTaskPool: new “shared” task pool subclass with slightly
+ different default behaviour than the existing GstTaskPool which
+ would create unlimited number of threads for new tasks. The shared
+ taskpool creates up to N threads (default: 1) and then distributes
+ pending tasks to those threads round-robin style, and blocks if no
+ thread is available. It is possible to join tasks. This can be used
+ by plugins to implement simple multi-threaded processing and is used
+ for the new multi-threaded video conversion and compositing done in
+ GstVideoAggregator, videoconverter and compositor.
+
+Plugins Base Utils library
+
+- GstDiscoverer:
+
+ - gst_discoverer_container_info_get_tags() was added to retrieve
+ global/container tags (vs. per-stream tags). Per-Stream tags can
+ be retrieved via the existing
+ gst_discoverer_stream_info_get_tags().
+ gst_discoverer_info_get_tags(), which for many files returns a
+ confusing mix of stream and container tags, has been deprecated
+ in favour of the container/stream-specific functions.
+ - gst_discoverer_stream_info_get_stream_number() returns a unique
+ integer identifier for a given stream within the given
+ GstDiscoverer context. (If this matches the stream number inside
+ the container bitstream that’s by coincidence and not by
+ design.)
+
+- gst_pb_utils_get_caps_description_flags() can be used to query
+ whether certain caps represent a container, audio, video, image,
+ subtitles, tags, or something else. This only works for formats
+ known to GStreamer.
+
+- gst_pb_utils_get_file_extension_from_caps() returns a possible file
+ extension for given caps.
+
+- gst_codec_utils_h264_get_profile_flags_level(): Parses profile,
+ flags, and level from H264 AvcC codec_data. The format of H264 AVCC
+ extradata/sequence_header is documented in the ITU-T H.264
+ specification section 7.3.2.1.1 as well as in ISO/IEC 14496-15
+ section 5.3.3.1.2.
+
+- gst_codec_utils_caps_get_mime_codec() to convert caps to a RFC 6381
+ compatible MIME codec string codec. Useful for providing the codecs
+ field inside the Content-Type HTTP header for containerized formats,
+ such as mp4 or matroska.
+
+GStreamer OpenGL integration library and plugins
+
+- glcolorconvert: added suppport for converting the video formats
+ A420, AV12, BGR, BGRA, RGBP and BGRP.
+
+- Added support to GstGLBuffer for persistent buffer mappings where a
+ Pixel Buffer Object (PBO) can be mapped by both the CPU and the GPU.
+ This removes a memcpy() when uploading textures or vertices
+ particularly when software decoders (e.g. libav) are direct
+ rendering into our memory. Improves transfer performance
+ significantly. Requires OpenGL 4.4, GL_ARB_buffer_storage or
+ GL_EXT_buffer_storage
+
+- Added various helper functions for handling 4x4 matrices of affine
+ transformations as used by GstVideoAffineTransformationMeta.
+
+- Add support to GstGLContext for allowing the application to control
+ the config (EGLConfig, GLXConfig, etc) used when creating the OpenGL
+ context. This allows the ability to choose between RGB16 or RGB10A2
+ or RGBA8 back/front buffer configurations that were previously
+ hardcoded. GstGLContext also supports retrieving the configuration
+ it was created with or from an externally provide OpenGL context
+ handle. This infrastructure is also used to create a compatible
+ config from an application/externally provided OpenGL context in
+ order to improve compatibility with other OpenGL frameworks and GUI
+ toolkits. A new environment variable GST_GL_CONFIG was also added to
+ be able to request a specific configuration from the command line.
+ Note: different platforms will have different functionality
+ available.
+
+- Add support for choosing between EGL and WGL at runtime when running
+ on Windows. Previously this was a build-time switch. Allows use in
+ e.g. Gtk applications on Windows that target EGL/ANGLE without
+ recompiling GStreamer. gst_gl_display_new_with_type() can be used by
+ applications to choose a specific display type to use.
+
+- Build fixes to explicitly check for Broadcom-specific libraries on
+ older versions of the Raspberry Pi platform. The Broadcom OpenGL ES
+ and EGL libraries have different filenames. Using the vc4 Mesa
+ driver on the Raspberry Pi is not affected.
+
+- Added support to glupload and gldownload for transferring RGBA
+ buffers using the memory:NVMM available on the Nvidia Tegra family
+ of embedded devices.
+
+- Added support for choosing libOpenGL and libGLX as used in a GLVND
+ environment on unix-based platforms. This allows using desktop
+ OpenGL and EGL without pulling in any GLX symbols as would be
+ required with libGL.
+
+Video library
+
+- New raw video formats:
+
+ - AV12 (NV12 with alpha plane)
+ - RGBP and BGRP (planar RGB formats)
+ - ARGB64 variants with specified endianness instead of host
+ endianness:
+ - ARGB64_LE, ARGB64_BE
+ - RGBA64_BE, RGBA64_LE
+ - BGRA64_BE, BGRA64_LE
+ - ABGR64_BE, ABGR64_LE
+
+- gst_video_orientation_from_tag() is new convenience API to parse the
+ image orientation from a GstTagList.
+
+- GstVideoDecoder subframe support (see below)
+
+- GstVideoCodecState now also carries some HDR metadata
+
+- Ancillary video data: implement transform functions for AFD/Bar
+ metas, so they will be forwarded in more cases
+
+MPEG-TS library
+
+This library only handles section parsing and such, see above for
+changes to the actual mpegtsmux and mpegtsdemux elements.
+
+- many additions and improvements to SCTE-35 section parsing
+- new API for fetching extended descriptors:
+ gst_mpegts_find_descriptor_with_extension()
+- add support for SIT sections (Selection Information Tables)
+- expose event-from-section constructor gst_event_new_mpegts_section()
+- parse Audio Preselection Descriptor needed for Dolby AC-4
+
+GstWebRTC library + webrtcbin
+
+- Change the way in which sink pads and transceivers are matched
+ together to support easier usage. If a pad is created without a
+ specific index (i.e. using sink_%u as the pad template), then an
+ available compatible transceiver will be searched for. If a specific
+ index is requested (i.e. sink_1) then if a transceiver for that
+ m-line already exists, that transceiver must match the new sink pad
+ request. If there is no transceiver available in either scenario, a
+ new transceiver is created. If a mixture of both sink_1 and sink_%u
+ requests result in an impossible situation, an error will be
+ produced at pad request time or from create offer/answer.
+
+- webrtcbin now uses regular ICE nomination instead of libnice’s
+ default of aggressive ICE nomination. Regular ICE nomination is the
+ default recommended by various relevant standards and improves
+ connectivity in specific network scenarios.
+
+- Add support for limiting the port range used for RTP with the
+ addition of the min-rtp-port and max-rtp-port properties on the ICE
+ object.
+
+- Expose the SCTP transport as a property on webrtcbin to more closely
+ match the WebRTC specification.
+
+- Added support for taking into account the data channel transport
+ state when determining the value of the "connection-state" property.
+ Previous versions of the WebRTC spec did not include the data
+ channel state when computing this value.
+
+- Add configuration for choosing the size of the underlying sockets
+ used for transporting media data
+
+- Always advertise support for the transport-cc RTCP feedback protocol
+ as rtpbin supports it. For full support, the configured caps (input
+ or through codec-preferences) need to include the relevant RTP
+ header extension.
+
+- Numerous fixes to caps and media handling to fail-fast when an
+ incompatible situation is detected.
+
+- Improved support for attaching the required media after a remote
+ offer has been set.
+
+- Add support for dynamically changing the amount of FEC used for a
+ particular stream.
+
+- webrtcbin now stops further SDP processing at the first error it
+ encounters.
+
+- Completed support for either local or the remote closing a data
+ channel.
+
+- Various fixes when performing BUNDLEing of the media streams in
+ relation to RTX and FEC usage.
+
+- Add support for writing out QoS DSCP marking on outgoing packets to
+ improve reliability in some network scenarios.
+
+- Improvements to the statistics returned by the get-stats signal
+ including the addition of the raw statistics from the internal
+ RTPSource, the TWCC stats when available.
+
+- The webrtc library does not expose any objects anymore with public
+ fields. Instead properties have been added to replace that
+ functionality. If you are accessing such fields in your application,
+ switch to the corresponding properties.
+
+GstCodecs and Video Parsers
+
+- Support for render delays to improve throughput across all CODECs
+ (used with NVDEC and V4L2).
+- lots of improvements to parsers and the codec parsing decoder base
+ classes (H264, H265, VP8, VP9, AV1, MPEG-2) used for various
+ hardware-accelerate decoder APIs.
+
+Bindings support
+
+- gst_allocation_params_new() allocates a GstAllocationParams struct
+ on the heap. This should only be used by bindings (and freed via
+ gst_allocation_params_free() then). In C code you would allocate
+ this on the stack and only init it in place.
+
+- gst_debug_log_literal() can be used to log a string to the debug log
+ without going through any printf format expansion and associated
+ overhead. This is mostly useful for bindings such as the Rust
+ bindings which may have done their own formatting already .
+
+- Provide non-inlined versions of refcounting APIs for various
+ GStreamer mini objects, so that they can be consumed by bindings
+ (e.g. gstreamer-sharp): gst_buffer_ref, gst_buffer_unref,
+ gst_clear_buffer, gst_buffer_copy, gst_buffer_replace,
+ gst_buffer_list_ref, gst_buffer_list_unref, gst_clear_buffer_list,
+ gst_buffer_list_copy, gst_buffer_list_replace, gst_buffer_list_take,
+ gst_caps_ref, gst_caps_unref, gst_clear_caps, gst_caps_replace,
+ gst_caps_take, gst_context_ref, gst_context_unref, gst_context_copy,
+ gst_context_replace, gst_event_replace, gst_event_steal,
+ gst_event_take, gst_event_ref, gst_event_unref, gst_clear_event,
+ gst_event_copy, gst_memory_ref, gst_memory_unref, gst_message_ref,
+ gst_message_unref, gst_clear_message, gst_message_copy,
+ gst_message_replace, gst_message_take, gst_promise_ref,
+ gst_promise_unref, gst_query_ref, gst_query_unref, gst_clear_query,
+ gst_query_copy, gst_query_replace, gst_query_take, gst_sample_ref,
+ gst_sample_unref, gst_sample_copy, gst_tag_list_ref,
+ gst_tag_list_unref, gst_clear_tag_list, gst_tag_list_replace,
+ gst_tag_list_take, gst_uri_copy, gst_uri_ref, gst_uri_unref,
+ gst_clear_uri.
+
+- expose a GType for GstMiniObject
+
+- gst_device_provider_probe() now returns non-floating device object
+
+API Deprecations
+
+- gst_element_get_request_pad() has been deprecated in favour of the
+ newly-added gst_element_request_pad_simple() which does the exact
+ same thing but has a less confusing name that hopefully makes clear
+ that the function request a new pad rather than just retrieves an
+ already-existing request pad.
+
+- gst_discoverer_info_get_tags(), which for many files returns a
+ confusing mix of stream and container tags, has been deprecated in
+ favour of the container-specific and stream-specific functions,
+ gst_discoverer_container_info_get_tags() and
+ gst_discoverer_stream_info_get_tags().
+
+- gst_video_sink_center_rect() was deprecated in favour of the more
+ generic newly-added gst_video_center_rect().
+
+- The GST_MEMORY_FLAG_NO_SHARE flag has been deprecated, as it tends
+ to cause problems and prevents sub-buffering. If pooling or lifetime
+ tracking is required, memories should be allocated through a custom
+ GstAllocator instead of relying on the lifetime of the buffers the
+ memories were originally attached to, which is fragile anyway.
+
+- The GstPlayer high-level playback library is being replaced with the
+ new GstPlay library (see above). GstPlayer should be considered
+ deprecated at this point and will be marked as such in the next
+ development cycle. Applications should be ported to GstPlay.
+
+- Gstreamer Editing Services: ges_video_transition_set_border(),
+ ges_video_transition_get_border()
+ ges_video_transition_set_inverted()
+ ges_video_transition_is_inverted() have been deprecated, use
+ ges_timeline_element_set_children_properties() instead.
Miscellaneous performance, latency and memory optimisations
-- this section will be filled in in due course
+More video conversion fast paths
-Miscellaneous other changes and enhancements
+- v210 ↔ I420, YV12, Y42B, UYVY and YUY2
+- A420 → RGB
+
+Less jitter when waiting on the system clock
+
+- Better system clock wait accuracy, less jitter: where available,
+ clock_nanosleep is used for higher accuracy for waits below 500
+ usecs, and waits below 2ms will first use the regular waiting system
+ and then clock_nanosleep for the remainder. The various wait
+ implementation have a latency ranging from 50 to 500+ microseconds.
+ While this is not a major issue when dealing with a low number of
+ waits per second (for ex: video), it does introduce a non-negligible
+ jitter for synchronisation of higher packet rate systems.
+
+Video decoder subframe support
-- this section will be filled in in due course
+- The GstVideoDecoder base class gained API to process input at the
+ sub-frame level. That way video decoders can start decoding slices
+ before they have received the full input frame in its entirety (to
+ the extent this is supported by the codec, of course). This helps
+ with CPU utilisation and reduces latency.
-Tracing framework and debugging improvements
+- This functionality is now being used in the OpenJPEG JPEG 2000
+ decoder, the FFmpeg H.264 decoder (in case of NAL-aligned input) and
+ the OpenMAX H.264/H.265 decoders (in case of NAL-aligned input).
-- this section will be filled in in due course
+Miscellaneous other changes and enhancements
+
+- GstDeviceMonitor no longer fails to start just because one of the
+ device providers failed to start. That could happen for example on
+ systems where the pulseaudio device provider is installed, but
+ pulseaudio isn’t actually running but ALSA is used for audio
+ instead. In the same vein the device monitor now keeps track of
+ which providers have been started (via the new
+ gst_device_provider_is_started()) and only stops actually running
+ device providers when stopping the device monitor.
+
+- On embedded systems it can be useful to create a registry that can
+ be shared and read by multiple processes running as different users.
+ It is now possible to set the new GST_REGISTRY_MODE environment
+ variable to specify the file mode for the registry file, which by
+ default is set to be only user readable/writable.
+
+- GstNetClientClock will signal lost sync in case the remote time
+ resets (e.g. because device power cycles), by emitting the “synced”
+ signal with synced=FALSE parameter, so applications can take action.
+
+- gst_value_deserialize_with_pspec() allows deserialization with a
+ hint for what the target GType should be. This allows for example
+ passing arrays of flags through the command line or
+ gst_util_set_object_arg(), eg: foo="<bar,bar+baz>".
+
+- It’s now allowed to create an empty GstVideoOverlayComposition
+ without any rectangles by passing a NULL rectangle to
+ gst_video_overlay_composition_new(). This is useful for bindings and
+ simplifies application code in some places.
+
+Tracing framework, debugging and testing improvements
+
+- New factories tracer to list loaded elements (and other plugin
+ features). This can be useful to collect a list of elements needed
+ for an application, which then in turn can be used to create a
+ tailored minimal GStreamer build that contains just the elements
+ needed and nothing else.
+- New plugin-feature-loaded tracing hook for use by tracers like the
+ new factories tracer
+
+- GstHarness: Add gst_harness_set_live() so that harnesses can be set
+ to non-live and return is-live=false in latency queries if needed.
+ Default behaviour is to always return is-live=true in latency
+ queries.
+
+- navseek: new "hold-eos" property. When enabled, the element will
+ hold back an EOS event until the next keystroke (via navigation
+ events). This can be used to keep a video sink showing the last
+ frame of a video pipeline until a key is pressed instead of tearing
+ it down immediately on EOS.
+
+- New fakeaudiosink element: mimics an audio sink and can be used for
+ testing and CI pipelines on systems where no audio system is
+ installed or running. It differs from fakesink in that it only
+ support audio caps and syncs to the clock by default like a normal
+ audio sink. It also implements the GstStreamVolume interface like
+ most audio sinks do.
+
+- New videocodectestsink element for video codec conformance testing:
+ Calculates MD5 checksums for video frames and skips any padding
+ whilst doing so. Can optionally also write back the video data with
+ padding removed into a file for easy byte-by-byte comparison with
+ reference data.
Tools
-- this section will be filled in in due course
+gst-inspect-1.0
+
+- Can sort the list of plugins by passing --sort=name as command line
+ option
+
+gst-launch-1.0
+
+- will now error out on top-level properties that don’t exist and
+ which were silently ignored before
+- On Windows the high-resolution clock is enabled now, which provides
+ better clock and timer performance on Windows (see Windows section
+ below for more details).
+
+gst-play-1.0
+
+- New --start-position command line argument to start playback from
+ the specified position
+- Audio can be muted/unmuted in interactive mode by pressing the m
+ key.
+- On Windows the high-resolution clock is enabled now (see Windows
+ section below for more details)
+
+gst-device-monitor-1.0
+
+- New --include-hidden command line argument to also show “hidden”
+ device providers
+
+ges-launch-1.0
+
+- New interactive mode that allows seeking and such. Can be disabled
+ by passing the --no-interactive argument on the command line.
+- Option to forward tags
+- Allow using an existing clip to determine the rendering format (both
+ topology and profile) via new --profile-from command line argument.
GStreamer RTSP server
-- this section will be filled in in due course
+- GstRTSPMediaFactory gained API to disable RTCP
+ (gst_rtsp_media_factory_set_enable_rtcp(), "enable-rtcp" property).
+ Previously RTCP was always allowed for all RTSP medias. With this
+ change it is possible to disable RTCP completely, no matter if the
+ client wants to do RTCP or not.
+
+- Make a mount point of / work correctly. While not allowed by the
+ RTSP 2 spec, the RTSP 1 spec is silent on this and it is used in the
+ wild. It is now possible to use / as a mount path in
+ gst-rtsp-server, e.g. rtsp://example.com/ would work with this now.
+ Note that query/fragment parts of the URI are not necessarily
+ correctly handled, and behaviour will differ between various
+ client/server implementations; so use it if you must but don’t bug
+ us if it doesn’t work with third party clients as you’d hoped.
+
+- multithreading fixes (races, refcounting issues, deadlocks)
+
+- ONVIF audio backchannel fixes
+
+- ONVIF trick mode optimisations
+
+- rtspclientsink: new "update-sdp" signal that allows updating the SDP
+ before sending it to the server via ANNOUNCE. This can be used to
+ add additional metadata to the SDP, for example. The order and
+ number of medias must not be changed, however.
GStreamer VAAPI
-- this section will be filled in in due course
+- new AV1 decoder element (vaapiav1dec)
+
+- H264 decoder: handle stereoscopic 3D video with frame packing
+ arrangement SEI messages
+
+- H265 encoder: added Screen Content Coding extensions support
+
+- H265 decoder: gained MAIN_444_12 profile support (decoded to
+ Y412_LE), and 4:2:2 12-bits support (decoded to Y212_LE)
+
+- vaapipostproc: gained BT2020 color standard support
+
+- vaapidecode: now generates caps templates dynamically at runtime in
+ order to advertise actually supported caps instead of all
+ theoretically supported caps.
+
+- GST_VAAPI_DRM_DEVICE environment variable to force a specified DRM
+ device when a DRM display is used. It is ignored when other types of
+ displays are used. By default /dev/dri/renderD128 is used for DRM
+ display.
GStreamer OMX
-- this section will be filled in in due course
+- subframe support in H.264/H.265 decoders
GStreamer Editing Services and NLE
-- this section will be filled in in due course
+- framepositioner: new "operator" property to access blending modes in
+ the compositor
+- timeline: Implement snapping to markers
+- smart-mixer: Add support for d3d11compositor and glvideomixer
+- titleclip: add "draw-shadow" child property
+- ges:// URI support to define a timeline from a description.
+- command-line-formatter
+ - Add track management to timeline description
+ - Add keyframe support
+- ges-launch-1.0:
+ - Add an interactive mode where we can seek etc…
+ - Add option to forward tags
+ - Allow using an existing clip to determine the rendering format
+ (both topology and profile) via new --profile-from command line
+ argument.
+- Fix static build
GStreamer validate
-- this section will be filled in in due course
+- report: Add a way to force backtraces on reports even if not a
+ critical issue (GST_VALIDATE_ISSUE_FLAGS_FORCE_BACKTRACE)
+- Add a flag to gst_validate_replace_variables_in_string() allow
+ defining how to resolve variables in structs
+- Add gst_validate_bin_monitor_get_scenario() to get the bin monitor
+ scenario, which is useful for applications that use Validate
+ directly.
+- Add an expected-values parameter to wait, message-type=XX allowing
+ more precise filtering of the message we are waiting for.
+- Add config file support: each test can now use a config file for the
+ given media file used to test.
+- Add support to check properties of object properties
+- scenario: Add an "action-done" signal to signal when an action is
+ done
+- scenario: Add a "run-command" action type
+- scenario: Allow forcing running action on idle from scenario file
+- scenario: Allow iterating over arrays in foreach
+- scenario: Rename ‘interlaced’ action to ‘non-blocking’
+- scenario: Add a non-blocking flag to the wait signal
GStreamer Python Bindings
-- this section will be filled in in due course
+- Fixes for Python 3.10
+- Various build fixes
+- at least one known breaking change caused by g-i annotation changes
+ (see below)
GStreamer C# Bindings
-- this section will be filled in in due course
+- Fix GstDebugGraphDetails enum
+- Updated to latests GtkSharp
+- Updated to include GStreamer 1.20 API
GStreamer Rust Bindings and Rust Plugins
-The GStreamer Rust bindings are released separately with a different
-release cadence that’s tied to gtk-rs, but the latest release has
-already been updated for the upcoming new GStreamer 1.20 API.
-
-gst-plugins-rs, the module containing GStreamer plugins written in Rust,
-has also seen lots of activity with many new elements and plugins.
-
-What follows is a list of elements and plugins available in
-gst-plugins-rs, so people don’t miss out on all those potentially useful
-elements that have no C equivalent.
-
-- FIXME: add new elements
-
-Rust audio plugins
-
-- audiornnoise: New element for audio denoising which implements the
- noise removal algorithm of the Xiph RNNoise library, in Rust
-- rsaudioecho: Port of the audioecho element from gst-plugins-good
- rsaudioloudnorm: Live audio loudness normalization element based on
- the FFmpeg af_loudnorm filter
-- claxondec: FLAC lossless audio codec decoder element based on the
- pure-Rust claxon implementation
-- csoundfilter: Audio filter that can use any filter defined via the
- Csound audio programming language
-- lewtondec: Vorbis audio decoder element based on the pure-Rust
- lewton implementation
-
-Rust video plugins
-
-- cdgdec/cdgparse: Decoder and parser for the CD+G video codec based
- on a pure-Rust CD+G implementation, used for example by karaoke CDs
-- cea608overlay: CEA-608 Closed Captions overlay element
-- cea608tott: CEA-608 Closed Captions to timed-text (e.g. VTT or SRT
- subtitles) converter
-- tttocea608: CEA-608 Closed Captions from timed-text converter
-- mccenc/mccparse: MacCaption Closed Caption format encoder and parser
-- sccenc/sccparse: Scenarist Closed Caption format encoder and parser
-- dav1dec: AV1 video decoder based on the dav1d decoder implementation
- by the VLC project
-- rav1enc: AV1 video encoder based on the fast and pure-Rust rav1e
- encoder implementation
-- rsflvdemux: Alternative to the flvdemux FLV demuxer element from
- gst-plugins-good, not feature-equivalent yet
-- rsgifenc/rspngenc: GIF/PNG encoder elements based on the pure-Rust
- implementations by the image-rs project
-
-Rust text plugins
-
-- textwrap: Element for line-wrapping timed text (e.g. subtitles) for
- better screen-fitting, including hyphenation support for some
- languages
-
-Rust network plugins
-
-- reqwesthttpsrc: HTTP(S) source element based on the Rust
- reqwest/hyper HTTP implementations and almost feature-equivalent
- with the main GStreamer HTTP source souphttpsrc
-- s3src/s3sink: Source/sink element for the Amazon S3 cloud storage
-- awstranscriber: Live audio to timed text transcription element using
- the Amazon AWS Transcribe API
-
-Generic Rust plugins
-
-- sodiumencrypter/sodiumdecrypter: Encryption/decryption element based
- on libsodium/NaCl
-- togglerecord: Recording element that allows to pause/resume
- recordings easily and considers keyframe boundaries
-- fallbackswitch/fallbacksrc: Elements for handling potentially
- failing (network) sources, restarting them on errors/timeout and
- showing a fallback stream instead
-- threadshare: Set of elements that provide alternatives for various
- existing GStreamer elements but allow to share the streaming threads
- between each other to reduce the number of threads
-- rsfilesrc/rsfilesink: File source/sink elements as replacements for
- the existing filesrc/filesink elements
+- The GStreamer Rust bindings are released separately with a different
+ release cadence that’s tied to gtk-rs, but the latest release has
+ already been updated for the upcoming new GStreamer 1.20 API (v1_20
+ feature).
+
+- gst-plugins-rs, the module containing GStreamer plugins written in
+ Rust, has also seen lots of activity with many new elements and
+ plugins. See the New Elements section above for a list of new Rust
+ elements.
Build and Dependencies
- Meson 0.59 or newer is required to build GStreamer now.
-- FIXME: this section will be filled in in due course
+- The GLib requirement has been bumped to GLib 2.56 or newer (from
+ March 2018).
+
+- The wpe plugin now requires wpe >= 2.28 and wpebackend-fdo >= 1.8
Explicit opt-in required for build of certain plugins with (A)GPL dependencies
See Building plugins with (A)GPL-licensed dependencies for more details
and a non-exhaustive list of plugins affected.
-gst-build: replaced by Monorepo
+gst-build: replaced by mono repository
-- this section will be filled in in due course
-
-- FIXME: describe + link to Monorepo FAQ
+See mono repository section above and the GStreamer mono repository FAQ.
Cerbero
on platforms where dependencies are not readily available, such as
Windows, Android, iOS and macOS.
-General improvements
+General Cerbero improvements
-- this section will be filled in in due course
+- Plugin removed: libvisual
+- New plugins: rtpmanagerbad and rist
-macOS / iOS
+macOS / iOS specific Cerbero improvements
-- this section will be filled in in due course
+- XCode 12 support
+- macOS OS release support is now future-proof, similar to iOS
+- macOS Apple Silicon (ARM64) cross-compile support has been added
+- macOS Apple Silicon (ARM64) native support is currently experimental
-Windows
+Windows specific Cerbero improvements
-- this section will be filled in in due course
+- Visual Studio 2022 support has been added
+- bootstrap is faster since it requires building fewer build-tools
+ recipes on Windows
+- package is faster due to better scheduling of recipe stages and
+ elimination of unnecessary autotools regeneration
+- The following plugins are no longer built on Windows:
+ - a52dec (another decoder is still available in libav)
+ - dvdread
+ - resindvd
Windows MSI installer
-- this section will be filled in in due course
+- no major changes
-Linux
+Linux specific Cerbero improvements
-- this section will be filled in in due course
+- Fedora, Debian OS release support is now more future-proof
+- Amazon Linux 2 support has been added
-Android
+Android specific Cerbero improvements
-- this section will be filled in in due course
+- no major changes
Platform-specific changes and improvements
Android
-- this section will be filled in in due course
+- No major changes
macOS and iOS
-- this section will be filled in in due course
+- applemedia: add ProRes support to vtenc and vtdec
Windows
-- this section will be filled in in due course
+- On Windows the high-resolution clock is enabled now in the
+ gst-launch-1.0 and gst-play-1.0 command line tools, which provides
+ better clock and timer performance on Windows, at the cost of higher
+ power consumption. By default, without the high-resolution clock
+ enabled, the timer precision on Windows is system-dependent and may
+ be as bad as 15ms which is not good enough for many multimedia
+ applications. Developers may want to do the same in their Windows
+ applications if they think it’s a good idea for their application
+ use case, and depending on the Windows version they target. This is
+ not done automatically by GStreamer because on older Windows
+ versions (pre-Windows 10) this affects a global Windows setting and
+ also there’s a power consumption vs. performance trade-off that may
+ differ from application to application.
+
+- dxgiscreencapsrc now supports resolution changes
+
+- The wasapi2 audio plugin was rewritten and now has a higher rank
+ than the old wasapi plugin since it has a number of additional
+ features such as automatic stream routing, and no
+ known-but-hard-to-fix issues. The plugin is always built if the
+ Windows 10 SDK is available now.
+
+- The wasapi device providers now detect and notify dynamic device
+ additions/removals
+
+- d3d11screencapturesrc: new desktop capture element, including
+ GstDeviceProvider implementation to enumerate/select target monitors
+ for capture.
+
+- Direct3D11/DXVA decoder now supports AV1 and MPEG2 codecs
+ (d3d11av1dec, d3d11mpeg2dec)
+
+- VP9 decoding got more reliable and stable thanks to a newly written
+ codec parser
+
+- Support for decoding interlaced H.264/AVC streams
+
+- Hardware-accelerated video deinterlacing (d3d11deinterlace) and
+ video mixing (d3d11compositor)
+
+- Video mixing with the Direct3D11 API (d3d11compositor)
+
+- MediaFoundation API based hardware encoders gained the ability to
+ receive Direct3D11 textures as an input
+
+- Seungha’s blog post “GStreamer ❤ Windows: A primer on the cool stuff
+ you’ll find in the 1.20 release” describes many of the
+ Windows-related improvements in more detail
Linux
-- this section will be filled in in due course
+- bluez: LDAC Bluetooth audio codec support in a2dpsink and avdtpsink,
+ as well as an LDAC RTP payloader (rtpldacpay) and an LDAC audio
+ encoder (ldacenc)
+
+- kmssink: gained support for NV24, NV61, RGB16/BGR16 formats;
+ auto-detect NVIDIA Tegra driver
Documentation improvements
-- this section will be filled in in due course
+- hardware-accelerated GPU plugins will now no longer always list all
+ the element variants for all available GPUs, since those are
+ system-dependent and it’s confusing for users to see those in the
+ documentation just because the GStreamer developer who generated the
+ docs had multiple GPUs to play with at the time. Instead just show
+ the default elements.
-Possibly Breaking Changes
+Possibly Breaking and Other Noteworthy Behavioural Changes
+
+- gst_parse_launch(), gst_parse_bin_from_description() and friends
+ will now error out when setting properties that don’t exist on
+ top-level bins. They were silently ignored before.
+
+- The GstWebRTC library does not expose any objects anymore with
+ public fields. Instead properties have been added to replace that
+ functionality. If you are accessing such fields in your application,
+ switch to the corresponding properties.
-- FIXME: this section will be filled in in due course
-- MPEG-TS SCTE-35 API changes (FIXME: flesh out)
-- gst_parse_launch() and friends now error out on non-existing
- properties on top-level bins where they would silently fail and
- ignore those before.
- playbin and uridecodebin now emit the source-setup signal before the
element is added to the bin and linked so that the source element is
already configured before any scheduling query comes in, which is
- useful for elements such as appsrc or giostreamsrc. (Merge Request)
+ useful for elements such as appsrc or giostreamsrc.
+
+- The source element inside urisourcebin (used inside uridecodebin3
+ which is used inside playbin3) is no longer called "source". This
+ shouldn’t affect anyone hopefully, because there’s a "setup-source"
+ signal to configure the source element and no one should rely on
+ names of internal elements anyway.
+
+- The vp8enc element now expects bps (bits per second) for the
+ "temporal-scalability-target-bitrate" property, which is consistent
+ with the "target-bitrate" property. Since additional configuration
+ is required with modern libvpx to make temporal scaling work anyway,
+ chances are that very few people will have been using this property
+
+- vp8enc and vp9enc now default to “good quality” for the "deadline"
+ property rather then “best quality”. Having the deadline set to best
+ quality causes the encoder to be absurdly slow, most real-life users
+ will want the good quality tradeoff instead.
+
+- The experimental GstTranscoder library API in gst-plugins-bad was
+ changed from a GObject signal-based notification mechanism to a
+ GstBus/message-based mechanism akin to GstPlayer/GstPlay.
+
+- MPEG-TS SCTE-35 API: semantic change for SCTE-35 splice commands:
+ timestamps passed by the application should be in running time now,
+ since users of the API can’t really be expected to predict the local
+ PTS of the muxer.
+
+- The GstContext used by souphttpsrc to share the session between
+ multiple element instances has changed. Previously it provided
+ direct access to the internal SoupSession object, now it only
+ provides access to an opaque, internal type. This change is
+ necessary because SoupSession is not thread-safe at all and can’t be
+ shared safely between arbitrary external code and souphttpsrc.
+
+- Python bindings: GObject-introspection related Annotation fixes have
+ led to a case of a GstVideo.VideoInfo-related function signature
+ changing in the Python bindings (possibly one or two other cases
+ too). This is for a function that should never have been exposed in
+ the first place though, so the bindings are being updated to throw
+ an exception in that case, and the correct replacement API has been
+ added in form of an override.
Known Issues
-- this section will be filled in in due course
-
-- There are a couple of known WebRTC-related regressions/blockers:
-
- - webrtc: DTLS setup with Chrome is broken
- - webrtcbin: First keyframe is usually lost
+- nothing in particular at this point (but also see possibly breaking
+ changes section above)
Contributors
-- this section will be filled in in due course
+Aaron Boxer, Adam Leppky, Adam Williamson, Alba Mendez, Alejandro
+González, Aleksandr Slobodeniuk, Alexander Vandenbulcke, Alex Ashley,
+Alicia Boya García, Andika Triwidada, Andoni Morales Alastruey, Andrew
+Wesie, Andrey Moiseev, Antonio Ospite, Antonio Rojas, Arthur Crippa
+Búrigo, Arun Raghavan, Ashley Brighthope, Axel Kellermann, Baek, Bastien
+Nocera, Bastien Reboulet, Benjamin Gaignard, Bing Song, Binh Truong,
+Biswapriyo Nath, Brad Hards, Brad Smith, Brady J. Garvin, Branko
+Subasic, Camilo Celis Guzman, Chris Bass, ChrisDuncanAnyvision, Chris
+White, Corentin Damman, Daniel Almeida, Daniel Knobe, Daniel Stone,
+david, David Fernandez, David Keijser, David Phung, Devarsh Thakkar,
+Dinesh Manajipet, Dmitry Samoylov, Dmitry Shusharin, Dominique Martinet,
+Doug Nazar, Ederson de Souza, Edward Hervey, Emmanuel Gil Peyrot,
+Enrique Ocaña González, Ezequiel Garcia, Fabian Orccon, Fabrice
+Fontaine, Fernando Jimenez Moreno, Florian Karydes, Francisco Javier
+Velázquez-García, François Laignel, Frederich Munch, Fredrik Pålsson,
+George Kiagiadakis, Georg Lippitsch, Göran Jönsson, Guido Günther,
+Guillaume Desmottes, Guiqin Zou, Haakon Sporsheim, Haelwenn (lanodan)
+Monnier, Haihao Xiang, Haihua Hu, Havard Graff, He Junyan, Helmut
+Januschka, Henry Wilkes, Hosang Lee, Hou Qi, Ignacio Casal Quinteiro,
+Igor Kovalenko, Ilya Kreymer, Imanol Fernandez, Jacek Tomaszewski, Jade
+Macho, Jakub Adam, Jakub Janků, Jan Alexander Steffens (heftig), Jan
+Schmidt, Jason Carrete, Jason Pereira, Jay Douglass, Jeongki Kim, Jérôme
+Laheurte, Jimmi Holst Christensen, Johan Sternerup, John Hassell, John
+Lindgren, John-Mark Bell, Jonathan Matthew, Jordan Petridis, Jose
+Quaresma, Julian Bouzas, Julien, Kai Uwe Broulik, Kasper Steensig
+Jensen, Kellermann Axel, Kevin Song, Khem Raj, Knut Inge Hvidsten, Knut
+Saastad, Kristofer Björkström, Lars Lundqvist, Lawrence Troup, Lim Siew
+Hoon, Lucas Stach, Ludvig Rappe, Luis Paulo Fernandes de Barros, Luke
+Yelavich, Mads Buvik Sandvei, Marc Leeman, Marco Felsch, Marek Vasut,
+Marian Cichy, Marijn Suijten, Marius Vlad, Markus Ebner, Mart Raudsepp,
+Matej Knopp, Mathieu Duponchelle, Matthew Waters, Matthieu De Beule,
+Mengkejiergeli Ba, Michael de Gans, Michael Olbrich, Michael Tretter,
+Michal Dzik, Miguel Paris, Mikhail Fludkov, mkba, Nazar Mokrynskyi,
+Nicholas Jackson, Nicola Murino, Nicolas Dufresne, Niklas Hambüchen,
+Nikolay Sivov, Nirbheek Chauhan, Olivier Blin, Olivier Crete, Olivier
+Crête, Paul Goulpié, Per Förlin, Peter Boba, P H, Philippe Normand,
+Philipp Zabel, Pieter Willem Jordaan, Piotrek Brzeziński, Rafał
+Dzięgiel, Rafostar, raghavendra, Raghavendra, Raju Babannavar, Raleigh
+Littles III, Randy Li, Randy Li (ayaka), Ratchanan Srirattanamet, Raul
+Tambre, reed.lawrence, Ricky Tang, Robert Rosengren, Robert Swain, Robin
+Burchell, Roman Sivriver, R S Nikhil Krishna, Ruben Gonzalez, Ruslan
+Khamidullin, Sanchayan Maity, Scott Moreau, Sebastian Dröge, Sergei
+Kovalev, Seungha Yang, Sid Sethupathi, sohwan.park, Sonny Piers, Staz M,
+Stefan Brüns, Stéphane Cerveau, Stephan Hesse, Stian Selnes, Stirling
+Westrup, Théo MAILLART, Thibault Saunier, Tim, Timo Wischer, Tim-Philipp
+Müller, Tim Schneider, Tobias Ronge, Tom Schoonjans, Tulio Beloqui,
+tyler-aicradle, U. Artie Eoff, Ung, Val Doroshchuk, VaL Doroshchuk,
+Víctor Manuel Jáquez Leal, Vivek R, Vivia Nikolaidou, Vivienne
+Watermeier, Vladimir Menshakov, Will Miller, Wim Taymans, Xabier
+Rodriguez Calvar, Xavier Claessens, Xℹ Ruoyao, Yacine Bandou, Yinhang
+Liu, youngh.lee, youngsoo.lee, yychao, Zebediah Figura, Zhang yuankun,
+Zhang Yuankun, Zhao, Zhao Zhili, , Aleksandar Topic, Antonio Ospite,
+Bastien Nocera, Benjamin Gaignard, Brad Hards, Carlos Falgueras García,
+Célestin Marot, Corentin Damman, Corentin Noël, Daniel Almeida, Daniel
+Knobe, Danny Smith, Dave Piché, Dmitry Osipenko, Fabrice Fontaine,
+fjmax, Florian Zwoch, Guillaume Desmottes, Haihua Hu, Heinrich Kruger,
+He Junyan, Jakub Adam, James Cowgill, Jan Alexander Steffens (heftig),
+Jean Felder, Jeongki Kim, Jiri Uncovsky, Joe Todd, Jordan Petridis,
+Krystian Wojtas, Marc-André Lureau, Marcin Kolny, Marc Leeman, Mark
+Nauwelaerts, Martin Reboredo, Mathieu Duponchelle, Matthew Waters,
+Mengkejiergeli Ba, Michael Gruner, Nicolas Dufresne, Nirbheek Chauhan,
+Olivier Crête, Philippe Normand, Rafał Dzięgiel, Ralf Sippl, Robert
+Mader, Sanchayan Maity, Sangchul Lee, Sebastian Dröge, Seungha Yang,
+Stéphane Cerveau, Teh Yule Kim, Thibault Saunier, Thomas Klausner, Timo
+Wischer, Tim-Philipp Müller, Tobias Reineke, Tomasz Andrzejak, Trung Do,
+Tyler Compton, Ung, Víctor Manuel Jáquez Leal, Vivia Nikolaidou, Wim
+Taymans, wngecn, Wonchul Lee, wuchang li, Xavier Claessens, Xi Ruoyao,
+Yoshiharu Hirose, Zhao,
… and many others who have contributed bug reports, translations, sent
suggestions or helped testing.
1.20.0
-1.20.0 is scheduled to be released around October/November 2021.
+1.20.0 is scheduled to be released around early February 2022.
Schedule for 1.22
unstable development version leading up to the stable 1.22 release. The
development of 1.21/1.22 will happen in the git main branch.
-The plan for the 1.22 development cycle is yet to be confirmed.
+The plan for the 1.22 development cycle is yet to be confirmed. Assuming
+no major project-wide reorganisations in the 1.22 cycle we might try and
+aim for a release around August 2022.
1.22 will be backwards-compatible to the stable 1.20, 1.18, 1.16, 1.14,
1.12, 1.10, 1.8, 1.6, 1.4, 1.2 and 1.0 release series.
------------------------------------------------------------------------
These release notes have been prepared by Tim-Philipp Müller with
-contributions from …
+contributions from Matthew Waters, Nicolas Dufresne, Nirbheek Chauhan,
+Sebastian Dröge and Seungha Yang.
License: CC BY-SA 4.0
-This is GStreamer gst-python 1.19.3.
+This is GStreamer gst-python 1.19.90.
GStreamer 1.19 is the development branch leading up to the next major
stable version which will be 1.20.
<release>
<Version>
+ <revision>1.19.90</revision>
+ <branch>main</branch>
+ <name></name>
+ <created>2022-01-28</created>
+ <file-release rdf:resource="https://gstreamer.freedesktop.org/src/gst-python/gst-python-1.19.90.tar.xz" />
+ </Version>
+ </release>
+
+ <release>
+ <Version>
<revision>1.19.3</revision>
<branch>main</branch>
<name></name>
project('gst-python', 'c', 'cpp',
- version : '1.19.3.1',
+ version : '1.19.90',
meson_version : '>= 0.59',
default_options : [ 'warning_level=1',
'c_std=gnu99',
+=== release 1.19.90 ===
+
+2022-01-28 14:28:35 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * NEWS:
+ * RELEASE:
+ * docs/gst_plugins_cache.json:
+ * gst-rtsp-server.doap:
+ * meson.build:
+ Release 1.19.90
+
+2022-01-28 14:28:28 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * ChangeLog:
+ Update ChangeLogs for 1.19.90
+
2022-01-20 17:13:36 -0600 Michael Gruner <michael.gruner@ridgerun.com>
* examples/test-appsrc2.c:
GStreamer 1.20 Release Notes
GStreamer 1.20 has not been released yet. It is scheduled for release in
-November 2021.
+late January / early February 2022.
1.19.x is the unstable development version that is being developed in
-the git main branch and which will eventually result in 1.20, and 1.19.3
-is the current development release in that series
-
-It is expected that feature freeze will be in early October 2021,
-followed by one or two 1.19.9x pre-releases and the new 1.20 stable
-release around October/November 2021.
+the git main branch and which will eventually result in 1.20, and
+1.19.90 is the first release candidate in that series (1.20rc1).
1.20 will be backwards-compatible to the stable 1.18, 1.16, 1.14, 1.12,
-1.10, 1.8, 1.6,, 1.4, 1.2 and 1.0 release series.
+1.10, 1.8, 1.6, 1.4, 1.2 and 1.0 release series.
See https://gstreamer.freedesktop.org/releases/1.20/ for the latest
version of this document.
-Last updated: Monday 1 November 2021, 01:00 UTC (log)
+Last updated: Wednesday 26 January 2022, 01:00 UTC (log)
Introduction
Highlights
-- this section will be completed in due course
+- Development in GitLab was switched to a single git repository
+ containing all the modules
+- GstPlay: new high-level playback library, replaces GstPlayer
+- WebM Alpha decoding support
+- Encoding profiles can now be tweaked with additional
+ application-specified element properties
+- Compositor: multi-threaded video conversion and mixing
+- RTP header extensions: unified support in RTP depayloader and
+ payloader base classes
+- SMPTE 2022-1 2-D Forward Error Correction support
+- Smart encoding (passthrough) support for VP8, VP9, H.265 in
+ encodebin and transcodebin
+- Runtime compatibility support for libsoup2 and libsoup3 (libsoup3
+ support experimental)
+- Video decoder subframe support
+- Video decoder automatic packet-loss, data corruption, and keyframe
+ request handling for RTP / WebRTC / RTSP
+- MP4 and Matroska muxers now support profile/level/resolution changes
+ for H264/H265 input streams (i.e. codec data changing on the fly)
+- MP4 muxing mode that initially creates a fragmented mp4 which is
+ converted to a regular mp4 on EOS
+- Audio support for the WebKit Port for Embedded (WPE) web page source
+ element
+- CUDA based video color space convert and rescale elements and
+ upload/download elements
+- NVIDIA memory:NVMM support for OpenGL glupload and gldownload
+ elements
+- Many WebRTC improvements
+- The new VA-API plugin implemention fleshed out with more decoders
+ and new postproc elements
+- AppSink API to retrieve events in addition to buffers and buffer
+ lists
+- AppSrc gained more configuration options for the internal queue
+ (leakiness, limits in buffers and time, getters to read current
+ levels)
+- Updated Rust bindings and many new Rust plugins
+- Improved support for custom minimal GStreamer builds
+- Support build against FFmpeg 5.0
+- Linux Stateless CODEC support gained MPEG2 and VP9
+- Windows Direct3D11/DXVA decoder gained AV1 and MPEG2 support
+- Lots of new plugins, features, performance improvements and bug
+ fixes
Major new features and changes
Noteworthy new features and API
-- this section will be filled in in due course
+- gst_element_get_request_pad() has been deprecated in favour of the
+ newly-added gst_element_request_pad_simple() which does the exact
+ same thing but has a less confusing name that hopefully makes clear
+ that the function request a new pad rather than just retrieves an
+ already-existing request pad.
+
+Development in GitLab was switched to a single git repository containing all the modules
+
+The GStreamer multimedia framework is a set of libraries and plugins
+split into a number of distinct modules which are released independently
+and which have so far been developed in separate git repositories in
+freedesktop.org GitLab.
+
+In addition to these separate git repositories there was a gst-build
+module that would use the Meson build systems’s subproject feature to
+download each individual module and then build everything in one go. It
+would also provide an uninstalled development environment that made it
+easy to work on GStreamer and use or test versions other than the
+system-installed GStreamer version.
+
+All of these modules have now (as of 28 September 2021) been merged into
+a single git repository (“Mono repository” or “monorepo”) which should
+simplify development workflows and continuous integration, especially
+where changes need to be made to multiple modules at once.
+
+This mono repository merge will primarily affect GStreamer developers
+and contributors and anyone who has workflows based on the GStreamer git
+repositories.
+
+The Rust bindings and Rust plugins modules have not been merged into the
+mono repository at this time because they follow a different release
+cycle.
+
+The mono repository lives in the existing GStreamer core git repository
+in GitLab in the new main branch and all future development will happen
+on this branch.
+
+Modules will continue to be released as separate tarballs.
+
+For more details, please see the GStreamer mono repository FAQ.
+
+GstPlay: new high-level playback library replacing GstPlayer
+
+- GstPlay is a new high-level playback library that replaces the older
+ GstPlayer API. It is basically the same API as GstPlayer but
+ refactored to use bus messages for application notifications instead
+ of GObject signals. There is still a signal adapter object for those
+ who prefer signals. Since the existing GstPlayer API is already in
+ use in various applications, it didn’t seem like a good idea to
+ break it entirely. Instead a new API was added, and it is expected
+ that this new GstPlay API will be moved to gst-plugins-base in
+ future.
+
+- The existing GstPlayer API is scheduled for deprecation and will be
+ removed at some point in the future (e.g. in GStreamer 1.24), so
+ application developers are urged to migrate to the new GstPlay API
+ at their earliest convenience.
+
+WebM alpha decoding
+
+- Implement WebM alpha decoding (VP8/VP9 with alpha), which required
+ support and additions in various places. This is supported both with
+ software decoders and hardware-accelerated decoders.
+
+- VP8/VP9 don’t support alpha components natively in the codec, so the
+ way this is implemented in WebM is by encoding the alpha plane with
+ transparency data as a separate VP8/VP9 stream. Inside the WebM
+ container (a variant of Matroska) this is coded as a single video
+ track with the “normal” VP8/VP9 video data making up the main video
+ data and each frame of video having an encoded alpha frame attached
+ to it as extra data ("BlockAdditional").
+
+- matroskademux has been extended extract this per-frame alpha side
+ data and attach it in form of a GstVideoCodecAlphaMeta to the
+ regular video buffers. Note that this new meta is specific to this
+ VP8/VP9 alpha support and can’t be used to just add alpha support to
+ other codecs that don’t support it. Lastly, matroskademux also
+ advertises the fact that the streams contain alpha in the caps.
+
+- The new codecalpha plugin contains various bits of infrastructure to
+ support autoplugging and debugging:
+
+ - codecalphademux splits out the alpha stream from the metas on
+ the regular VP8/VP9 buffers
+ - alphacombine takes two decoded raw video streams (one alpha, one
+ the regular video) and combines it into a video stream with
+ alpha
+ - vp8alphadecodebin + vp9alphadecodebin are wrapper bins that use
+ the regular vp8dec and vp9dec software decoders to decode
+ regular and alpha streams and combine them again. To decodebin
+ these look like regular decoders which ju
+ - The V4L2 CODEC plugin has stateless VP8/VP9 decoders that can
+ decode both alpha and non-alpha stream with a single decoder
+ instance
+
+- A new AV12 video format was added which is basically NV12 with an
+ alpha plane, which is more convenient for many hardware-accelerated
+ decoders.
+
+- Watch Nicolas Dufresne’s LCA 2022 talk “Bringing WebM Alpha support
+ to GStreamer” for all the details and a demo.
+
+RTP Header Extensions Base Class and Automatic Header Extension Handling in RTP Payloaders and Depayloaders
+
+- RTP Header Extensions are specified in RFC 5285 and provide a way to
+ add small pieces of data to RTP packets in between the RTP header
+ and the RTP payload. This is often used for per-frame metadata,
+ extended timestamps or other application-specific extra data. There
+ are several commonly-used extensions specified in various RFCs, but
+ senders are free to put any kind of data in there, as long as sender
+ and receiver both know what that data is. Receivers that don’t know
+ about the header extensions will just skip the extra data without
+ ever looking at it. These header extensions can often be combined
+ with any kind of payload format, so may need to be supported by many
+ RTP payloader and depayloader elements.
+
+- Inserting and extracting RTP header extension data has so far been a
+ bit inconvenient in GStreamer: There are functions to add and
+ retrieve RTP header extension data from RTP packets, but nothing
+ works automatically, even for common extensions. People would have
+ to do the insertion/extraction either in custom elements
+ before/after the RTP payloader/depayloader, or inside pad probes,
+ which isn’t very nice.
+
+- This release adds various pieces of new infrastructure for generic
+ RTP header extension handling, as well as some implementations for
+ common extensions:
+
+ - GstRTPHeaderExtension is a new helper base class for reading and
+ writing RTP header extensions. Nominally this subclasses
+ GstElement, but only so these extensions are stored in the
+ registry where they can be looked up by URI or name. They don’t
+ have pads and don’t get added to the pipeline graph as an
+ element.
+
+ - "add-extension" and "clear-extension" action signals on RTP
+ payloaders and depayloaders for manual extension management
+
+ - The "request-extension" signal will be emitted if an extension
+ is encountered that requires explicit mapping by the application
+
+ - new "auto-header-extension" property on RTP payloaders and
+ depayloaders for automatic handling of known header extensions.
+ This is enabled by default. The extensions must be signalled via
+ caps / SDP.
+
+ - RTP header extension implementations:
+
+ - rtphdrextclientaudiolevel: Client-to-Mixer Audio Level
+ Indication (RFC 6464) (also see below)
+ - rtphdrextcolorspace: Color Space extension, extends RTP
+ packets with color space and high dynamic range (HDR)
+ information
+ - rtphdrexttwcc: Transport Wide Congestion Control support
+
+- gst_rtp_buffer_remove_extension_data() is a new helper function to
+ remove an RTP header extension from an RTP buffer
+
+- The existing gst_rtp_buffer_set_extension_data() now also supports
+ shrinking the extension data in size
+
+AppSink and AppSrc improvements
+
+- appsink: new API to pull events out of appsink in addition to
+ buffers and buffer lists.
+
+ There was previously no way for users to receive incoming events
+ from appsink properly serialised with the data flow, even if they
+ are serialised events. The reason for that is that the only way to
+ intercept events was via a pad probe on the appsink sink pad, but
+ there is also internal queuing inside of appsink, so it’s difficult
+ to ascertain the right order of everything in all cases.
+
+ There is now a new "new-serialized-event" signal which will be
+ emitted when there’s a new event pending (just like the existing
+ "new-sample" signal). The "emit-signals" property must be set to
+ TRUE in order to activate this (but it’s also fine to just pull from
+ the application thread without using the signals).
+
+ gst_app_sink_pull_object() and gst_app_sink_try_pull_object() can be
+ used to pull out either an event or a new sample carrying a buffer
+ or buffer list, whatever is next in the queue.
+
+ EOS events will be filtered and will not be returned. EOS handling
+ can be done the ususal way, same as with _pull_sample().
+
+- appsrc: allow configuration of internal queue limits in time and
+ buffers and add leaky mode.
+
+ There is internal queuing inside appsrc so the application thread
+ can push data into the element which will then be picked up by the
+ source element’s streaming thread and pushed into the pipeline from
+ that streaming thread. This queue is unlimited by default and until
+ now it was only possible to set a maximum size limit in bytes. When
+ that byte limit is reached, the pushing thread (application thread)
+ would be blocked until more space becomes available.
+
+ A limit in bytes is not particularly useful for many use cases, so
+ now it is possible to also configure limits in time and buffers
+ using the new "max-time" and "max-buffers" properties. Of course
+ there are also matching new read-only"current-level-buffers" and
+ "current-level-time properties" properties to query the current fill
+ level of the internal queue in time and buffers.
+
+ And as if that wasn’t enough the internal queue can also be
+ configured as leaky using the new "leaky-type" property. That way
+ when the queue is full the application thread won’t be blocked when
+ it tries to push in more data, but instead either the new buffer
+ will be dropped or the oldest data in the queue will be dropped.
+
+Better string serialization of nested GstCaps and GstStructures
+
+- New string serialisation format for structs and caps that can handle
+ nested structs and caps properly by using brackets to delimit nested
+ items (e.g. some-struct, some-field=[nested-struct, nested=true]).
+ Unlike the default format the new variant can also support more than
+ one level of nesting. For backwards-compatibility reasons the old
+ format is still output by default when serialising caps and structs
+ using the existing API. The new functions gst_caps_serialize() and
+ gst_structure_serialize() can be used to output strings in the new
+ format.
+
+Convenience API for custom GstMetas
+
+- New convenience API to register and create custom GstMetas:
+ gst_meta_register_custom() and gst_buffer_add_custom_meta(). Such
+ custom meta is backed by a GstStructure and does not require that
+ users of the API expose their GstMeta implementation as public API
+ for other components to make use of it. In addition, it provides a
+ simpler interface by ignoring the impl vs. api distinction that the
+ regular API exposes. This new API is meant to be the meta
+ counterpart to custom events and messages, and to be more convenient
+ than the lower-level API when the absolute best performance isn’t a
+ requirement. The reason it’s less performant than a “proper” meta is
+ that a proper meta is just a C struct in the end whereas this goes
+ through the GstStructure API which has a bit more overhead, which
+ for most scenarios is negligible however. This new API is useful for
+ experimentation or proprietary metas, but also has some limitations:
+ it can only be used if there’s a single producer of these metas;
+ it’s not allowed to register the same custom meta multiple times or
+ from multiple places.
+
+Additional Element Properties on Encoding Profiles
+
+- GstEncodingProfile: The new "element-properties" and
+ gst_encoding_profile_set_element_properties() API allows
+ applications to set additional element properties on encoding
+ profiles to configure muxers and encoders. So far the encoding
+ profile template was the only place where this could be specified,
+ but often what applications want to do is take a ready-made encoding
+ profile shipped by GStreamer or the application and then tweak the
+ settings on top of that, which is now possible with this API. Since
+ applications can’t always know in advance what encoder element will
+ be used in the end, it’s even possible to specify properties on a
+ per-element basis.
+
+ Encoding Profiles are used in the encodebin, transcodebin and
+ camerabin elements and APIs to configure output formats (containers
+ and elementary streams).
+
+Audio Level Indication Meta for RFC 6464
+
+- New GstAudioLevelMeta containing Audio Level Indication as per RFC
+ 6464
+
+- The level element has been updated to add GstAudioLevelMeta on
+ buffers if the "audio-level-meta" property is set to TRUE. This can
+ then in turn be picked up by RTP payloaders to signal the audio
+ level to receivers through RTP header extensions (see above).
+
+- New Client-to-Mixer Audio Level Indication (RFC6464) RTP Header
+ Extension which should be automatically created and used by RTP
+ payloaders and depayloaders if their "auto-header-extension"
+ property is enabled and if the extension is part of the RTP caps.
+
+Automatic packet loss, data corruption and keyframe request handling for video decoders
+
+- The GstVideoDecoder base class has gained various new APIs to
+ automatically handle packet loss and data corruption better by
+ default, especially in RTP, RTSP and WebRTC streaming scenarios, and
+ to give subclasses more control about how they want to handle
+ missing data:
+
+ - Video decoder subclasses can mark output frames as corrupted via
+ the new GST_VIDEO_CODEC_FRAME_FLAG_CORRUPTED flag
+
+ - A new "discard-corrupted-frames" property allows applications to
+ configure decoders so that corrupted frames are directly
+ discarded instead of being forwarded inside the pipeline. This
+ is a replacement for the "output-corrupt" property of the FFmpeg
+ decoders.
+
+ - RTP depayloaders can now signal to decoders that data is missing
+ when sending GAP events for lost packets. GAP events can be sent
+ for various reason in a GStreamer pipeline. Often they are just
+ used to let downstream elements know that there isn’t a buffer
+ available at the moment, so downstream elements can move on
+ instead of waiting for one. They are also sent by RTP
+ depayloaders in the case that packets are missing, however, and
+ so far a decoder was not able to differentiate the two cases.
+ This has been remedied now: GAP events can be decorated with
+ gst_event_set_gap_flags() and GST_GAP_FLAG_MISSING_DATA to let
+ decoders now what happened, and decoders can then use that in
+ some cases to handle missing data better.
+
+ - The GstVideoDecoder::handle_missing_data vfunc was added to
+ inform subclasses about packet loss or missing data and let them
+ handle it in their own way if they like.
+
+ - gst_video_decoder_set_needs_sync_point() lets subclasses signal
+ that they need the stream to start with a sync point. If
+ enabled, the base class will discard all non-sync point frames
+ in the beginning and after a flush and does not pass them to the
+ subclass. Furthermore, if the first frame is not a sync point,
+ the base class will try and request a sync frame from upstream
+ by sending a force-key-unit event (see next items).
+
+ - New "automatic-request-sync-points" and
+ "automatic-request-sync-point-flags" properties to automatically
+ request sync points when needed, e.g. on packet loss or if the
+ first frame is not a keyframe. Applications may want to enable
+ this on decoders operating in e.g. RTP/WebRTC/RTSP receiver
+ pipelines.
+
+ - The new "min-force-key-unit-interval" property can be used to
+ ensure there’s a minimal interval between keyframe requests to
+ upstream (and/or the sender) and we’re not flooding the sender
+ with key unit requests.
+
+ - gst_video_decoder_request_sync_point() allows subclasses to
+ request a new sync point (e.g. if they choose to do their own
+ missing data handling). This will still honour the
+ "min-force-key-unit-interval" property if set.
+
+Improved support for custom minimal GStreamer builds
+
+- Element registration and registration of other plugin features
+ inside plugin init functions has been improved in order to
+ facilitate minimal custom GStreamer builds.
+
+- A number of new macros have been added to declare and create
+ per-element and per-pluginfeature register functions in all plugins,
+ and then call those from the per-plugin plugin_init functions:
+
+ - GST_ELEMENT_REGISTER_DEFINE,
+ GST_DEVICE_PROVIDER_REGISTER_DEFINE,
+ GST_DYNAMIC_TYPE_REGISTER_DEFINE, GST_TYPE_FIND_REGISTER_DEFINE
+ for the actual registration call with GStreamer
+ - GST_ELEMENT_REGISTER, GST_DEVICE_PROVIDER_REGISTER,
+ GST_DYNAMIC_TYPE_REGISTER, GST_PLUGIN_STATIC_REGISTER,
+ GST_TYPE_FIND_REGISTER to call the registration function defined
+ by the REGISTER_DEFINE macro
+ - GST_ELEMENT_REGISTER_DECLARE,
+ GST_DEVICE_PROVIDER_REGISTER_DECLARE,
+ GST_DYNAMIC_TYPE_REGISTER_DECLARE,
+ GST_TYPE_FIND_REGISTER_DECLARE to declare the registration
+ function defined by the REGISTER_DEFINE macro
+ - and various variants for advanced use cases.
+
+- This means that applications can call the per-element and
+ per-pluginfeature registration functions for only the elements they
+ need instead of registering plugins as a whole with all kinds of
+ elements that may not be required (e.g. encoder and decoder instead
+ of just decoder). In case of static linking all unused functions and
+ their dependencies would be removed in this case by the linker,
+ which helps minimise binary size for custom builds.
+
+- gst_init() will automatically call a gst_init_static_plugins()
+ function if one exists.
+
+- See the GStreamer static build documentation and Stéphane’s blog
+ post Generate a minimal GStreamer build, tailored to your needs for
+ more details.
New elements
-- this section will be filled in in due course
+- New aesdec and aesenc elements for AES encryption and decryption in
+ a custom format.
+
+- New encodebin2 element with dynamic/sometimes source pads in order
+ to support the option of doing the muxing outside of encodebin,
+ e.g. in combination with a splitmuxsink.
+
+- New fakeaudiosink and videocodectestsink elements for testing and
+ debugging (see below for more details)
+
+- rtpisacpay, rtpisacdepay: new RTP payloader and depayloader for iSAC
+ audio codec
+
+- rtpst2022-1-fecdec, rtpst2022-1-fecenc: new elements providing SMPTE
+ 2022-1 2-D Forward Error Correction. More details in Mathieu’s blog
+ post.
+
+- isac: new plugin wrapping the Internet Speech Audio Codec reference
+ encoder and decoder from the WebRTC project.
+
+- asio: plugin for Steinberg ASIO (Audio Streaming Input/Output) API
+
+- gssrc, gssink: add source and sink for Google Cloud Storage
+
+- onnx: new plugin to apply ONNX neural network models to video
+
+- openaptx: aptX and aptX-HD codecs using libopenaptx (v0.2.0)
+
+- qroverlay, debugqroverlay: new elements that allows overlaying data
+ on top of video in form of a QR code
+
+- cvtracker: new OpenCV-based tracker element
+
+- av1parse, vp9parse: new parsers for AV1 and VP9 video
+
+- va: work on the new VA-API plugin implementation for
+ hardware-accelerated video decoding and encoding has continued at
+ pace, with various new decoders and filters having joined the
+ initial vah264dec:
+
+ - vah265dec: VA-API H.265 decoder
+ - vavp8dec: VA-API VP8 decoder
+ - vavp9dec: VA-API VP9 decoder
+ - vaav1dec: VA-API AV1 decoder
+ - vampeg2dec: VA-API MPEG-2 decoder
+ - vadeinterlace: : VA-API deinterlace filter
+ - vapostproc: : VA-API postproc filter (color conversion,
+ resizing, cropping, color balance, video rotation, skin tone
+ enhancement, denoise, sharpen)
+
+ See Víctor’s blog post “GstVA in GStreamer 1.20” for more details
+ and what’s coming up next.
+
+- vaapiav1dec: new AV1 decoder element (in gstreamer-vaapi)
+
+- msdkav1dec: hardware-accelerated AV1 decoder using the Intel Media
+ SDK / oneVPL
+
+- nvcodec plugin for NVIDIA NVCODEC API for hardware-accelerated video
+ encoding and decoding:
+
+ - cudaconvert, cudascale: new CUDA based video color space convert
+ and rescale elements
+ - cudaupload, cudadownload: new helper elements for memory
+ transfer between CUDA and system memory spaces
+ - nvvp8sldec, nvvp9sldec: new GstCodecs-based VP8/VP9 decoders
+
+- Various new hardware-accelerated elements for Windows:
+
+ - d3d11screencapturesrc: new desktop capture element, including a
+ GstDeviceProvider implementation to enumerate/select target
+ monitors for capture.
+ - d3d11av1dec and d3d11mpeg2dec: AV1 and MPEG-2 decoders
+ - d3d11deinterlace: deinterlacing filter
+ - d3d11compositor: video composing element
+ - see Windows section below for more details
+
+- new Rust plugins:
+
+ - audiornnoise: Removes noise from an audio stream
+ - awstranscribeparse: Parses AWS audio transcripts into timed text
+ buffers
+ - ccdetect: Detects if valid closed captions are present in a
+ closed captions stream
+ - cea608tojson: Converts CEA-608 Closed Captions to a JSON
+ representation
+ - cmafmux: CMAF fragmented MP4 muxer
+ - dashmp4mux: DASH fragmented MP4 muxer
+ - isofmp4mux: ISO fragmented MP4 muxer
+ - ebur128level: EBU R128 Loudness Level Measurement
+ - ffv1dec: FFV1 video decoder
+ - gtk4paintablesink: GTK4 video sink, which provides a
+ GdkPaintable that can be rendered in various widgets
+ - hlssink3: HTTP Live Streaming sink
+ - hrtfrender: Head-Related Transfer Function (HRTF) renderer
+ - hsvdetector: HSV colorspace detector
+ - hsvfilter: HSV colorspace filter
+ - jsongstenc: Wraps buffers containing any valid top-level JSON
+ structures into higher level JSON objects, and outputs those as
+ ndjson
+ - jsongstparse: Parses ndjson as output by jsongstenc
+ - jsontovtt: converts JSON to WebVTT subtitles
+ - regex: Applies regular expression operations on text
+ - roundedcorners: Adds rounded corners to video
+ - spotifyaudiosrc: Spotify source
+ - textahead: Display upcoming text buffers ahead (e.g. for
+ Karaoke)
+ - transcriberbin: passthrough bin that transcribes raw audio to
+ closed captions using awstranscriber and puts the captions as
+ metas onto the video
+ - tttojson: Converts timed text to a JSON representation
+ - uriplaylistbin: Playlist source bin
+ - webpdec-rs: WebP image decoder with animation support
+
+- New plugin codecalpha with elements to assist with WebM Alpha
+ decoding
+
+ - codecalphademux: Split stream with GstVideoCodecAlphaMeta into
+ two streams
+ - alphacombine: Combine two raw video stream (I420 or NV12) as one
+ stream with alpha channel (A420 or AV12)
+ - vp8alphadecodebin: A bin to handle software decoding of VP8 with
+ alpha
+ - vp9alphadecodebin: A bin to handle software decoding of VP9 with
+ alpha
+
+- New hardware accelerated elements for Linux:
+
+ - v4l2slmpeg2dec: Support for Linux Stateless MPEG2 decoders
+ - v4l2slvp9dec: Support for Linux Stateless VP9 decoders
+ - v4l2slvp8alphadecodebin: Support HW accelerated VP8 with alpha
+ layer decoding
+ - v4l2slvp9alphadecodebin: Support HW accelerated VP9 with alpha
+ layer decoding
New element features and additions
-- this section will be filled in in due course
+- assrender: handle more font mime types; better interaction with
+ matroskademux for embedded fonts
+
+- audiobuffersplit: Add support for specifying output buffer size in
+ bytes (not just duration)
+
+- audiolatency: new "samplesperbuffer" property so users can configure
+ the number of samples per buffer. The default value is 240 samples
+ which is equivalent to 5ms latency with a sample rate of 48000,
+ which might be larger than actual buffer size of audio capture
+ device.
+
+- audiomixer, audiointerleave, GstAudioAggregator: now keep a count of
+ samples that are dropped or processed as statistic and can be made
+ to post QoS messages on the bus whenever samples are dropped by
+ setting the "qos-messages" property on input pads.
+
+- audiomixer, compositor: improved handling of new inputs added at
+ runtime. New API was added to the GstAggregator base class to allow
+ subclasses to opt into an aggregation mode where inactive pads are
+ ignored when processing input buffers
+ (gst_aggregator_set_ignore_inactive_pads(),
+ gst_aggregator_pad_is_inactive()). An “inactive pad” in this context
+ is a pad which, in live mode, hasn’t yet received a first buffer,
+ but has been waited on at least once. What would happen usually in
+ this case is that the aggregator would wait for data on this pad
+ every time, up to the maximum configured latency. This would
+ inadvertently push mixer elements in live mode to the configured
+ latency envelope and delay processing when new inputs are added at
+ runtime until these inputs have actually produced data. This is
+ usually undesirable. With this new API, new inputs can be added
+ (requested) and configured and they won’t delay the data processing.
+ Applications can opt into this new behaviour by setting the
+ "ignore-inactive-pads" property on compositor, audiomixer or other
+ GstAudioAggregator-based elements.
+
+- cccombiner: implement “scheduling” of captions. So far cccombiner’s
+ behaviour was essentially that of a funnel: it strictly looked at
+ input timestamps to associate together video and caption buffers.
+ Now it will try to smoothly schedule caption buffers in order to
+ have exactly one per output video buffer. This might involve
+ rewriting input captions, for example when the input is CDP then
+ sequence counters are rewritten, time codes are dropped and
+ potentially re-injected if the input video frame had a time code
+ meta. This can also lead to the input drifting from synchronization,
+ when there isn’t enough padding in the input stream to catch up. In
+ that case the element will start dropping old caption buffers once
+ the number of buffers in its internal queue reaches a certain limit
+ (configurable via the "max-scheduled" property). The new original
+ funnel-like behaviour can be restored by setting the "scheduling"
+ property to FALSE.
+
+- ccconverter: new "cdp-mode" property to specify which sections to
+ include in CDP packets (timecode, CC data, service info). Various
+ software, including ffmpeg’s Decklink support, fails parsing CDP
+ packets that contain anything but CC data in the CDP packets.
+
+- clocksync: new "sync-to-first" property for automatic timestamp
+ offset setup: if set clocksync will set up the "ts-offset" value
+ based on the first buffer and the pipeline’s running time when the
+ first buffer arrived. The newly configured "ts-offset" in this case
+ would be the value that allows outputting the first buffer without
+ waiting on the clock. This is useful for example to feed a non-live
+ input into an already-running pipeline.
+
+- compositor:
+
+ - multi-threaded input conversion and compositing. Set the
+ "max-threads" property to activate this.
+ - new "sizing-policy" property to support display aspect ratio
+ (DAR)-aware scaling. By default the image is scaled to fill the
+ configured destination rectangle without padding and without
+ keeping the aspect ratio. With sizing-policy=keep-aspect-ratio
+ the input image is scaled to fit the destination rectangle
+ specified by GstCompositorPad:{xpos, ypos, width, height}
+ properties preserving the aspect ratio. As a result, the image
+ will be centered in the destination rectangle with padding if
+ necessary.
+ - new "zero-size-is-unscaled" property on input pads. By default
+ pad width=0 or pad height=0 mean that the stream should not be
+ scaled in that dimension. But if the "zero-size-is-unscaled"
+ property is set to FALSE a width or height of 0 is instead
+ interpreted to mean that the input image on that pad should not
+ be composited, which is useful when creating animations where an
+ input image is made smaller and smaller until it disappears.
+ - improved handling of new inputs at runtime via
+ "ignore-inactive-pads"property (see above for details)
+ - allow output format with alpha even if none of the inputs have
+ alpha (also glvideomixer and other GstVideoAggregator
+ subclasses)
+
+- dashsink: add h265 codec support and signals for allowing custom
+ playlist/fragment output
+
+- decodebin3:
+
+ - improved decoder selection, especially for hardware decoders
+ - make input activation “atomic” when adding inputs dynamically
+ - better interleave handling: take into account decoder latency
+ for interleave size
+
+- decklink:
+
+ - Updated DeckLink SDK to 11.2 to support DeckLink 8K Pro
+ - decklinkvideosrc:
+ - More accurate and stable capture timestamps: use the
+ hardware reference clock time when the frame was finished
+ being captured instead of a clock time much further down the
+ road.
+ - Automatically detect widescreen vs. normal NTSC/PAL
+
+- encodebin:
+
+ - add “smart encoding” support for H.265, VP8 and VP9 (i.e. only
+ re-encode where needed and otherwise pass through encoded video
+ as-is).
+ - H264/H265 smart encoding improvements: respect user-specified
+ stream-format, but if not specified default to avc3/hvc1 with
+ in-band SPS/PPS/VPS signalling for more flexibility.
+ - new encodebin2 element with dynamic/sometimes source pads in
+ order to support the option of doing the muxing outside of
+ encodebin, e.g. in combination with splitmuxsink.
+ - add APIs to set element properties on encoding profiles (see
+ below)
+
+- errorignore: new "ignore-eos" property to also ignore FLOW_EOS from
+ downstream elements
+
+- giosrc: add support for growing source files: applications can
+ specify that the underlying file being read is growing by setting
+ the "is-growing" property. If set, the source won’t EOS when it
+ reaches the end of the file, but will instead start monitoring it
+ and will start reading data again whenever a change is detected. The
+ new "waiting-data" and "done-waiting-data" signals keep the
+ application informed about the current state.
+
+- gtksink, gtkglsink:
+
+ - scroll event support: forwarded as navigation events into the
+ pipeline
+ - "video-aspect-ratio-override" property to force a specific
+ aspect ratio
+ - "rotate-method" property and support automatic rotation based on
+ image tags
+
+- identity: new "stats" property allows applications to retrieve the
+ number of bytes and buffers that have passed through so far.
+
+- interlace: add support for more formats, esp 10-bit, 12-bit and
+ 16-bit ones
+
+- jack: new "low-latency" property for automatic latency-optimized
+ setting and "port-names" property to select ports explicitly
+
+- jpegdec: support output conversion to RGB using libjpeg-turbo (for
+ certain input files)
+
+- line21dec:
+
+ - "mode" property to control whether and how detected closed
+ captions should be inserted in the list of existing close
+ caption metas on the input frame (if any): add, drop, or
+ replace.
+ - "ntsc-only" property to only look for captions if video has NTSC
+ resolution
+
+- line21enc: new "remove-caption-meta" to remove metas from output
+ buffers after encoding the captions into the video data; support for
+ CDP closed captions
+
+- matroskademux, matroskamux: Add support for ffv1, a lossless
+ intra-frame video coding format.
+
+- matroskamux: accept in-band SPS/PPS/VPS for H264 and H265
+ (i.e. stream-format avc3 and hev1) which allows on-the-fly
+ profile/level/resolution changes.
+
+- matroskamux: new "cluster-timestamp-offset" property, useful for use
+ cases where the container timestamps should map to some absolute
+ wall clock time, for example.
+
+- rtpsrc: add "caps" property to allow explicit setting of the caps
+ where needed
+
+- mpegts: support SCTE-35 passthrough via new "send-scte35-events"
+ property on MPEG-TS demuxer tsdemux. When enabled, SCTE 35 sections
+ (eg ad placement opportunities) are forwarded as events donwstream
+ where they can be picked up again by mpegtsmux. This required a
+ semantic change in the SCTE-35 section API: timestamps are now in
+ running time instead of muxer pts.
+
+- tsdemux: Handle PCR-less MPEG-TS streams; more robust timestamp
+ handling in certain corner cases and for poorly muxed streams.
+
+- mpegtsmux:
+
+ - More conformance improvements to make MPEG-TS analyzers happy:
+ - PCR timing accuracy: Improvements to the way mpegtsmux
+ outputs PCR observations in CBR mode, so that a PCR
+ observation is always inserted when needed, so that we never
+ miss the configured pcr-interval, as that triggers various
+ MPEG-TS analyser errors.
+ - Improved PCR/SI scheduling
+ - Don’t write PCR until PAT/PMT are output to make sure streams
+ start cleanly with a PAT/PMT.
+ - Allow overriding the automatic PMT PID selection via
+ application-supplied PMT_%d fields in the prog-map
+ structure/property.
+
+- mp4mux:
+
+ - new "first-moov-then-finalise" mode for fragmented output where
+ the output will start with a self-contained moov atom for the
+ first fragment, and then produce regular fragments. Then at the
+ end when the file is finalised, the initial moov is invalidated
+ and a new moov is written covering the entire file. This way the
+ file is a “fragmented mp4” file while it is still being written
+ out, and remains playable at all times, but at the end it is
+ turned into a regular mp4 file (with former fragment headers
+ remaining as unused junk data in the file).
+ - support H.264 avc3 and H.265 hvc1 stream formats as input where
+ the codec data is signalled in-band inside the bitstream instead
+ of caps/file headers.
+ - support profile/level/resolution changes for H264/H265 input
+ streams (i.e. codec data changing on the fly). Each codec_data
+ is put into its own SampleTableEntry inside the stsd, unless the
+ input is in avc3 stream format in which case it’s written
+ in-band an not in the headers.
+
+- multifilesink: new ""min-keyframe-distance"" property to make
+ minimum distance between keyframes in next-file=key-frame mode
+ configurable instead of hard-coding it to 10 seconds.
+
+- mxfdemux has seen a big refactoring to support non-frame wrappings
+ and more accurate timestamp/seek handling for some formats
+
+- msdk plugin for hardware-accelerated video encoding and decoding
+ using the Intel Media SDK:
+
+ - oneVPL support (Intel oneAPI Video Processing Library)
+ - AV1 decoding support
+ - H.264 decoder now supports constrained-high and progressive-high
+ profiles
+ - H.264 encoder:
+ - more configuration options (properties):
+ "intra-refresh-type", "min-qp" , "max-qp", "p-pyramid",
+ "dblk-idc"
+ - H.265 encoder:
+ - can output main-still-picture profile
+ - now inserts HDR SEIs (mastering display colour volume and
+ content light level)
+ - more configuration options (properties):
+ "intra-refresh-type", "min-qp" , "max-qp", "p-pyramid",
+ "b-pyramid", "dblk-idc", "transform-skip"
+ - support for RGB 10bit format
+ - External bitrate control in encoders
+ - Video post proc element msdkvpp gained support for 12-bit pixel
+ formats P012_LE, Y212_LE and Y412_LE
+
+- nvh264sldec: interlaced stream support
+
+- openh264enc: support main, high, constrained-high and
+ progressive-high profiles
+
+- openjpeg: support for multithreaded decoding and encoding
+
+- rtspsrc: now supports IPv6 also for tunneled mode (RTSP-over-HTTP);
+ new "ignore-x-server-reply" property to ignore the
+ x-server-ip-address server header reply in case of HTTP tunneling,
+ as it is often broken.
+
+- souphttpsrc: Runtime compatibility support for libsoup2 and
+ libsoup3. libsoup3 is the latest major version of libsoup, but
+ libsoup2 and libsoup3 can’t co-exist in the same process because
+ there is no namespacing or versioning for GObject types. As a
+ result, it would be awkward if the GStreamer souphttpsrc plugin
+ linked to a specific version of libsoup, because it would only work
+ with applications that use the same version of libsoup. To make this
+ work, the soup plugin now tries to determine the libsoup version
+ used by the application (and its other dependencies) at runtime on
+ systems where GStreamer is linked dynamically. libsoup3 support is
+ still considered somewhat experimental at this point.
+
+- srtsrc, srtsink: add signals for the application to accept/reject
+ incoming connections
+
+- timeoverlay: new elapsed-running-time time mode which shows the
+ running time since the first running time (and each flush-stop).
+
+- udpsrc: new timestamping mode to retrieve packet receive timestamps
+ from the kernel via socket control messages (SO_TIMESTAMPNS) on
+ supported platforms
+
+- uritranscodebin: new setup-source and element-setup signals for
+ applications to configure elements used
+
+- v4l2codecs plugin gained support for 4x4 and 32x32 tile formats
+ enabling some platforms or direct renders. Important memory usage
+ improvement.
+
+- v4l2slh264dec now implements the final Linux uAPI as shipped on
+ Linux 5.11 and later.
+
+- valve: add "drop-mode" property and provide two new modes of
+ operation: in drop-mode=forward-sticky-events sticky events
+ (stream-start, segment, tags, caps, etc.) are forwarded downstream
+ even when dropping is enabled; drop-mode=transform-to-gap will in
+ addition also convert buffers into gap events when dropping is
+ enabled, which lets downstream elements know that time is advancing
+ and might allow for preroll in many scenarios. By default all events
+ and all buffers are dropped when dropping is enabled, which can
+ cause problems with caps negotiation not progressing or branches not
+ prerolling when dropping is enabled.
+
+- videocrop: support for many more pixel formats, e.g. planar YUV
+ formats with > 8bits and GBR* video formats; can now also accept
+ video not backed by system memory as long as downstream supports the
+ GstCropMeta
+
+- videotestsrc: new smpte-rp-219 pattern for SMPTE75 RP-219 conformant
+ color bars
+
+- vp8enc: finish support for temporal scalability: two new properties
+ ("temporal-scalability-layer-flags",
+ "temporal-scalability-layer-sync-flags") and a unit change on the
+ "temporal-scalability-target-bitrate" property (now expects bps);
+ also make temporal scalability details available to RTP payloaders
+ as buffer metadata.
+
+- vp9enc: new properties to tweak encoder performance:
+
+ - "aq-mode" to configure adaptive quantization modes
+ - "frame-parallel-decoding" to configure whether to create a
+ bitstream that reduces decoding dependencies between frames
+ which allows staged parallel processing of more than one video
+ frames in the decoder. (Defaults to TRUE)
+ - "row-mt", "tile-columns" and "tile-rows" so multithreading can
+ be enabled on a per-tile basis, instead of on a per tile-column
+ basis. In combination with the new "tile-rows" property, this
+ allows the encoder to make much better use of the available CPU
+ power.
+
+- vp9dec, vp9enc: add support for 10-bit 4:2:0 and 4:2:2 YUV, as well
+ as 8-bit 4:4:4
+
+- vp8enc, vp9enc now default to “good quality” for the deadline
+ property rather then “best quality”. Having the deadline set to best
+ quality causes the encoder to be absurdly slow, most real-life users
+ will prefer good-enough quality with better performance instead.
+
+- wpesrc:
+
+ - implement audio support: a new sometimes source pad will be
+ created for each audio stream created by the web engine.
+ - move wpesrc to wpevideosrc and add a wrapper bin wpesrc to also
+ support audio
+ - also handles web:// URIs now (same as cefsrc)
+ - post messages with the estimated load progress on the bus
-Plugin and library moves
+- x265enc: add negative DTS support, which means timestamps are now
+ offset by 1h same as with x264enc
+
+RTP Payloaders and Depayloaders
+
+- rtpisacpay, rtpisacdepay: new RTP payloader and depayloader for iSAC
+ audio codec
+
+- rtph264depay:
+
+ - new "request-keyframe" property to make the depayloader
+ automatically request a new keyframe from the sender on packet
+ loss, consistent with the new property on rtpvp8depay.
+ - new "wait-for-keyframe" property to make depayloader wait for a
+ new keyframe at the beginning and after packet loss (only
+ effective if the depayloader outputs AUs), consistent with the
+ existing property on rtpvp8depay.
+
+- rtpopuspay, rtpopusdepay: support libwebrtc-compatible multichannel
+ audio in addition to the previously supported multichannel audio
+ modes
+
+- rtpopuspay: add DTX (Discontinuous Transmission) support
+
+- rtpvp8depay: new "request-keyframe" property to make the depayloader
+ automatically request a new keyframe from the sender on packet loss.
+
+- rtpvp8pay: temporal scaling support
-- this section will be filled in in due course
+- rtpvp9depay: Improved SVC handling (aggregate all layers)
+
+RTP Infrastructure
+
+- rtpst2022-1-fecdec, rtpst2022-1-fecenc: new elements providing SMPTE
+ 2022-1 2-D Forward Error Correction. More details in Mathieu’s blog
+ post.
+
+- rtpreddec: BUNDLE support
+
+- rtpredenc, rtpulpfecenc: add support for Transport-wide Congestion
+ Control (TWCC)
+
+- rtpsession: new "twcc-feedback-interval" property to allow RTCP TWCC
+ reports to be scheduled on a timer instead of per marker-bit.
+
+Plugin and library moves
- There were no plugin moves or library moves in this cycle.
The following elements or plugins have been removed:
-- this section will be filled in in due course
-
- The ofa audio fingerprinting plugin has been removed. The MusicIP
database has been defunct for years so this plugin is likely neither
useful nor used by anyone.
Miscellaneous API additions
-- this section will be filled in in due course
+Core
+
+- gst_buffer_new_memdup() is a convenience function for the
+ widely-used gst_buffer_new_wrapped(g_memdup(data,size),size)
+ pattern.
+
+- gst_caps_features_new_single() creates a new single GstCapsFeatures,
+ avoiding the need to use the vararg function with NULL terminator
+ for simple cases.
+
+- gst_element_type_set_skip_documentation() can be used by plugins to
+ signal that certain elements should not be included in the GStreamer
+ plugin documentation. This is useful for plugins where elements are
+ registered dynamically based on hardware capabilities and/or where
+ the available plugins and properties vary from system to system.
+ This is used in the d3d11 plugin for example to ensure that only the
+ list of default elements is advertised in the documentation.
+
+- gst_type_find_suggest_empty_simple() is a new convenience function
+ for typefinders for cases where there’s only a media type and no
+ other fields.
+
+- New API to create elements and set properties at construction time,
+ which is not only convenient, but also allows GStreamer elements to
+ have construct-only properties: gst_element_factory_make_full(),
+ gst_element_factory_make_valist(),
+ gst_element_factory_make_with_properties(),
+ gst_element_factory_create_full(),
+ gst_element_factory_create_valist(),
+ gst_element_factory_create_with_properties().
+
+- GstSharedTaskPool: new “shared” task pool subclass with slightly
+ different default behaviour than the existing GstTaskPool which
+ would create unlimited number of threads for new tasks. The shared
+ taskpool creates up to N threads (default: 1) and then distributes
+ pending tasks to those threads round-robin style, and blocks if no
+ thread is available. It is possible to join tasks. This can be used
+ by plugins to implement simple multi-threaded processing and is used
+ for the new multi-threaded video conversion and compositing done in
+ GstVideoAggregator, videoconverter and compositor.
+
+Plugins Base Utils library
+
+- GstDiscoverer:
+
+ - gst_discoverer_container_info_get_tags() was added to retrieve
+ global/container tags (vs. per-stream tags). Per-Stream tags can
+ be retrieved via the existing
+ gst_discoverer_stream_info_get_tags().
+ gst_discoverer_info_get_tags(), which for many files returns a
+ confusing mix of stream and container tags, has been deprecated
+ in favour of the container/stream-specific functions.
+ - gst_discoverer_stream_info_get_stream_number() returns a unique
+ integer identifier for a given stream within the given
+ GstDiscoverer context. (If this matches the stream number inside
+ the container bitstream that’s by coincidence and not by
+ design.)
+
+- gst_pb_utils_get_caps_description_flags() can be used to query
+ whether certain caps represent a container, audio, video, image,
+ subtitles, tags, or something else. This only works for formats
+ known to GStreamer.
+
+- gst_pb_utils_get_file_extension_from_caps() returns a possible file
+ extension for given caps.
+
+- gst_codec_utils_h264_get_profile_flags_level(): Parses profile,
+ flags, and level from H264 AvcC codec_data. The format of H264 AVCC
+ extradata/sequence_header is documented in the ITU-T H.264
+ specification section 7.3.2.1.1 as well as in ISO/IEC 14496-15
+ section 5.3.3.1.2.
+
+- gst_codec_utils_caps_get_mime_codec() to convert caps to a RFC 6381
+ compatible MIME codec string codec. Useful for providing the codecs
+ field inside the Content-Type HTTP header for containerized formats,
+ such as mp4 or matroska.
+
+GStreamer OpenGL integration library and plugins
+
+- glcolorconvert: added suppport for converting the video formats
+ A420, AV12, BGR, BGRA, RGBP and BGRP.
+
+- Added support to GstGLBuffer for persistent buffer mappings where a
+ Pixel Buffer Object (PBO) can be mapped by both the CPU and the GPU.
+ This removes a memcpy() when uploading textures or vertices
+ particularly when software decoders (e.g. libav) are direct
+ rendering into our memory. Improves transfer performance
+ significantly. Requires OpenGL 4.4, GL_ARB_buffer_storage or
+ GL_EXT_buffer_storage
+
+- Added various helper functions for handling 4x4 matrices of affine
+ transformations as used by GstVideoAffineTransformationMeta.
+
+- Add support to GstGLContext for allowing the application to control
+ the config (EGLConfig, GLXConfig, etc) used when creating the OpenGL
+ context. This allows the ability to choose between RGB16 or RGB10A2
+ or RGBA8 back/front buffer configurations that were previously
+ hardcoded. GstGLContext also supports retrieving the configuration
+ it was created with or from an externally provide OpenGL context
+ handle. This infrastructure is also used to create a compatible
+ config from an application/externally provided OpenGL context in
+ order to improve compatibility with other OpenGL frameworks and GUI
+ toolkits. A new environment variable GST_GL_CONFIG was also added to
+ be able to request a specific configuration from the command line.
+ Note: different platforms will have different functionality
+ available.
+
+- Add support for choosing between EGL and WGL at runtime when running
+ on Windows. Previously this was a build-time switch. Allows use in
+ e.g. Gtk applications on Windows that target EGL/ANGLE without
+ recompiling GStreamer. gst_gl_display_new_with_type() can be used by
+ applications to choose a specific display type to use.
+
+- Build fixes to explicitly check for Broadcom-specific libraries on
+ older versions of the Raspberry Pi platform. The Broadcom OpenGL ES
+ and EGL libraries have different filenames. Using the vc4 Mesa
+ driver on the Raspberry Pi is not affected.
+
+- Added support to glupload and gldownload for transferring RGBA
+ buffers using the memory:NVMM available on the Nvidia Tegra family
+ of embedded devices.
+
+- Added support for choosing libOpenGL and libGLX as used in a GLVND
+ environment on unix-based platforms. This allows using desktop
+ OpenGL and EGL without pulling in any GLX symbols as would be
+ required with libGL.
+
+Video library
+
+- New raw video formats:
+
+ - AV12 (NV12 with alpha plane)
+ - RGBP and BGRP (planar RGB formats)
+ - ARGB64 variants with specified endianness instead of host
+ endianness:
+ - ARGB64_LE, ARGB64_BE
+ - RGBA64_BE, RGBA64_LE
+ - BGRA64_BE, BGRA64_LE
+ - ABGR64_BE, ABGR64_LE
+
+- gst_video_orientation_from_tag() is new convenience API to parse the
+ image orientation from a GstTagList.
+
+- GstVideoDecoder subframe support (see below)
+
+- GstVideoCodecState now also carries some HDR metadata
+
+- Ancillary video data: implement transform functions for AFD/Bar
+ metas, so they will be forwarded in more cases
+
+MPEG-TS library
+
+This library only handles section parsing and such, see above for
+changes to the actual mpegtsmux and mpegtsdemux elements.
+
+- many additions and improvements to SCTE-35 section parsing
+- new API for fetching extended descriptors:
+ gst_mpegts_find_descriptor_with_extension()
+- add support for SIT sections (Selection Information Tables)
+- expose event-from-section constructor gst_event_new_mpegts_section()
+- parse Audio Preselection Descriptor needed for Dolby AC-4
+
+GstWebRTC library + webrtcbin
+
+- Change the way in which sink pads and transceivers are matched
+ together to support easier usage. If a pad is created without a
+ specific index (i.e. using sink_%u as the pad template), then an
+ available compatible transceiver will be searched for. If a specific
+ index is requested (i.e. sink_1) then if a transceiver for that
+ m-line already exists, that transceiver must match the new sink pad
+ request. If there is no transceiver available in either scenario, a
+ new transceiver is created. If a mixture of both sink_1 and sink_%u
+ requests result in an impossible situation, an error will be
+ produced at pad request time or from create offer/answer.
+
+- webrtcbin now uses regular ICE nomination instead of libnice’s
+ default of aggressive ICE nomination. Regular ICE nomination is the
+ default recommended by various relevant standards and improves
+ connectivity in specific network scenarios.
+
+- Add support for limiting the port range used for RTP with the
+ addition of the min-rtp-port and max-rtp-port properties on the ICE
+ object.
+
+- Expose the SCTP transport as a property on webrtcbin to more closely
+ match the WebRTC specification.
+
+- Added support for taking into account the data channel transport
+ state when determining the value of the "connection-state" property.
+ Previous versions of the WebRTC spec did not include the data
+ channel state when computing this value.
+
+- Add configuration for choosing the size of the underlying sockets
+ used for transporting media data
+
+- Always advertise support for the transport-cc RTCP feedback protocol
+ as rtpbin supports it. For full support, the configured caps (input
+ or through codec-preferences) need to include the relevant RTP
+ header extension.
+
+- Numerous fixes to caps and media handling to fail-fast when an
+ incompatible situation is detected.
+
+- Improved support for attaching the required media after a remote
+ offer has been set.
+
+- Add support for dynamically changing the amount of FEC used for a
+ particular stream.
+
+- webrtcbin now stops further SDP processing at the first error it
+ encounters.
+
+- Completed support for either local or the remote closing a data
+ channel.
+
+- Various fixes when performing BUNDLEing of the media streams in
+ relation to RTX and FEC usage.
+
+- Add support for writing out QoS DSCP marking on outgoing packets to
+ improve reliability in some network scenarios.
+
+- Improvements to the statistics returned by the get-stats signal
+ including the addition of the raw statistics from the internal
+ RTPSource, the TWCC stats when available.
+
+- The webrtc library does not expose any objects anymore with public
+ fields. Instead properties have been added to replace that
+ functionality. If you are accessing such fields in your application,
+ switch to the corresponding properties.
+
+GstCodecs and Video Parsers
+
+- Support for render delays to improve throughput across all CODECs
+ (used with NVDEC and V4L2).
+- lots of improvements to parsers and the codec parsing decoder base
+ classes (H264, H265, VP8, VP9, AV1, MPEG-2) used for various
+ hardware-accelerate decoder APIs.
+
+Bindings support
+
+- gst_allocation_params_new() allocates a GstAllocationParams struct
+ on the heap. This should only be used by bindings (and freed via
+ gst_allocation_params_free() then). In C code you would allocate
+ this on the stack and only init it in place.
+
+- gst_debug_log_literal() can be used to log a string to the debug log
+ without going through any printf format expansion and associated
+ overhead. This is mostly useful for bindings such as the Rust
+ bindings which may have done their own formatting already .
+
+- Provide non-inlined versions of refcounting APIs for various
+ GStreamer mini objects, so that they can be consumed by bindings
+ (e.g. gstreamer-sharp): gst_buffer_ref, gst_buffer_unref,
+ gst_clear_buffer, gst_buffer_copy, gst_buffer_replace,
+ gst_buffer_list_ref, gst_buffer_list_unref, gst_clear_buffer_list,
+ gst_buffer_list_copy, gst_buffer_list_replace, gst_buffer_list_take,
+ gst_caps_ref, gst_caps_unref, gst_clear_caps, gst_caps_replace,
+ gst_caps_take, gst_context_ref, gst_context_unref, gst_context_copy,
+ gst_context_replace, gst_event_replace, gst_event_steal,
+ gst_event_take, gst_event_ref, gst_event_unref, gst_clear_event,
+ gst_event_copy, gst_memory_ref, gst_memory_unref, gst_message_ref,
+ gst_message_unref, gst_clear_message, gst_message_copy,
+ gst_message_replace, gst_message_take, gst_promise_ref,
+ gst_promise_unref, gst_query_ref, gst_query_unref, gst_clear_query,
+ gst_query_copy, gst_query_replace, gst_query_take, gst_sample_ref,
+ gst_sample_unref, gst_sample_copy, gst_tag_list_ref,
+ gst_tag_list_unref, gst_clear_tag_list, gst_tag_list_replace,
+ gst_tag_list_take, gst_uri_copy, gst_uri_ref, gst_uri_unref,
+ gst_clear_uri.
+
+- expose a GType for GstMiniObject
+
+- gst_device_provider_probe() now returns non-floating device object
+
+API Deprecations
+
+- gst_element_get_request_pad() has been deprecated in favour of the
+ newly-added gst_element_request_pad_simple() which does the exact
+ same thing but has a less confusing name that hopefully makes clear
+ that the function request a new pad rather than just retrieves an
+ already-existing request pad.
+
+- gst_discoverer_info_get_tags(), which for many files returns a
+ confusing mix of stream and container tags, has been deprecated in
+ favour of the container-specific and stream-specific functions,
+ gst_discoverer_container_info_get_tags() and
+ gst_discoverer_stream_info_get_tags().
+
+- gst_video_sink_center_rect() was deprecated in favour of the more
+ generic newly-added gst_video_center_rect().
+
+- The GST_MEMORY_FLAG_NO_SHARE flag has been deprecated, as it tends
+ to cause problems and prevents sub-buffering. If pooling or lifetime
+ tracking is required, memories should be allocated through a custom
+ GstAllocator instead of relying on the lifetime of the buffers the
+ memories were originally attached to, which is fragile anyway.
+
+- The GstPlayer high-level playback library is being replaced with the
+ new GstPlay library (see above). GstPlayer should be considered
+ deprecated at this point and will be marked as such in the next
+ development cycle. Applications should be ported to GstPlay.
+
+- Gstreamer Editing Services: ges_video_transition_set_border(),
+ ges_video_transition_get_border()
+ ges_video_transition_set_inverted()
+ ges_video_transition_is_inverted() have been deprecated, use
+ ges_timeline_element_set_children_properties() instead.
Miscellaneous performance, latency and memory optimisations
-- this section will be filled in in due course
+More video conversion fast paths
-Miscellaneous other changes and enhancements
+- v210 ↔ I420, YV12, Y42B, UYVY and YUY2
+- A420 → RGB
+
+Less jitter when waiting on the system clock
+
+- Better system clock wait accuracy, less jitter: where available,
+ clock_nanosleep is used for higher accuracy for waits below 500
+ usecs, and waits below 2ms will first use the regular waiting system
+ and then clock_nanosleep for the remainder. The various wait
+ implementation have a latency ranging from 50 to 500+ microseconds.
+ While this is not a major issue when dealing with a low number of
+ waits per second (for ex: video), it does introduce a non-negligible
+ jitter for synchronisation of higher packet rate systems.
+
+Video decoder subframe support
-- this section will be filled in in due course
+- The GstVideoDecoder base class gained API to process input at the
+ sub-frame level. That way video decoders can start decoding slices
+ before they have received the full input frame in its entirety (to
+ the extent this is supported by the codec, of course). This helps
+ with CPU utilisation and reduces latency.
-Tracing framework and debugging improvements
+- This functionality is now being used in the OpenJPEG JPEG 2000
+ decoder, the FFmpeg H.264 decoder (in case of NAL-aligned input) and
+ the OpenMAX H.264/H.265 decoders (in case of NAL-aligned input).
-- this section will be filled in in due course
+Miscellaneous other changes and enhancements
+
+- GstDeviceMonitor no longer fails to start just because one of the
+ device providers failed to start. That could happen for example on
+ systems where the pulseaudio device provider is installed, but
+ pulseaudio isn’t actually running but ALSA is used for audio
+ instead. In the same vein the device monitor now keeps track of
+ which providers have been started (via the new
+ gst_device_provider_is_started()) and only stops actually running
+ device providers when stopping the device monitor.
+
+- On embedded systems it can be useful to create a registry that can
+ be shared and read by multiple processes running as different users.
+ It is now possible to set the new GST_REGISTRY_MODE environment
+ variable to specify the file mode for the registry file, which by
+ default is set to be only user readable/writable.
+
+- GstNetClientClock will signal lost sync in case the remote time
+ resets (e.g. because device power cycles), by emitting the “synced”
+ signal with synced=FALSE parameter, so applications can take action.
+
+- gst_value_deserialize_with_pspec() allows deserialization with a
+ hint for what the target GType should be. This allows for example
+ passing arrays of flags through the command line or
+ gst_util_set_object_arg(), eg: foo="<bar,bar+baz>".
+
+- It’s now allowed to create an empty GstVideoOverlayComposition
+ without any rectangles by passing a NULL rectangle to
+ gst_video_overlay_composition_new(). This is useful for bindings and
+ simplifies application code in some places.
+
+Tracing framework, debugging and testing improvements
+
+- New factories tracer to list loaded elements (and other plugin
+ features). This can be useful to collect a list of elements needed
+ for an application, which then in turn can be used to create a
+ tailored minimal GStreamer build that contains just the elements
+ needed and nothing else.
+- New plugin-feature-loaded tracing hook for use by tracers like the
+ new factories tracer
+
+- GstHarness: Add gst_harness_set_live() so that harnesses can be set
+ to non-live and return is-live=false in latency queries if needed.
+ Default behaviour is to always return is-live=true in latency
+ queries.
+
+- navseek: new "hold-eos" property. When enabled, the element will
+ hold back an EOS event until the next keystroke (via navigation
+ events). This can be used to keep a video sink showing the last
+ frame of a video pipeline until a key is pressed instead of tearing
+ it down immediately on EOS.
+
+- New fakeaudiosink element: mimics an audio sink and can be used for
+ testing and CI pipelines on systems where no audio system is
+ installed or running. It differs from fakesink in that it only
+ support audio caps and syncs to the clock by default like a normal
+ audio sink. It also implements the GstStreamVolume interface like
+ most audio sinks do.
+
+- New videocodectestsink element for video codec conformance testing:
+ Calculates MD5 checksums for video frames and skips any padding
+ whilst doing so. Can optionally also write back the video data with
+ padding removed into a file for easy byte-by-byte comparison with
+ reference data.
Tools
-- this section will be filled in in due course
+gst-inspect-1.0
+
+- Can sort the list of plugins by passing --sort=name as command line
+ option
+
+gst-launch-1.0
+
+- will now error out on top-level properties that don’t exist and
+ which were silently ignored before
+- On Windows the high-resolution clock is enabled now, which provides
+ better clock and timer performance on Windows (see Windows section
+ below for more details).
+
+gst-play-1.0
+
+- New --start-position command line argument to start playback from
+ the specified position
+- Audio can be muted/unmuted in interactive mode by pressing the m
+ key.
+- On Windows the high-resolution clock is enabled now (see Windows
+ section below for more details)
+
+gst-device-monitor-1.0
+
+- New --include-hidden command line argument to also show “hidden”
+ device providers
+
+ges-launch-1.0
+
+- New interactive mode that allows seeking and such. Can be disabled
+ by passing the --no-interactive argument on the command line.
+- Option to forward tags
+- Allow using an existing clip to determine the rendering format (both
+ topology and profile) via new --profile-from command line argument.
GStreamer RTSP server
-- this section will be filled in in due course
+- GstRTSPMediaFactory gained API to disable RTCP
+ (gst_rtsp_media_factory_set_enable_rtcp(), "enable-rtcp" property).
+ Previously RTCP was always allowed for all RTSP medias. With this
+ change it is possible to disable RTCP completely, no matter if the
+ client wants to do RTCP or not.
+
+- Make a mount point of / work correctly. While not allowed by the
+ RTSP 2 spec, the RTSP 1 spec is silent on this and it is used in the
+ wild. It is now possible to use / as a mount path in
+ gst-rtsp-server, e.g. rtsp://example.com/ would work with this now.
+ Note that query/fragment parts of the URI are not necessarily
+ correctly handled, and behaviour will differ between various
+ client/server implementations; so use it if you must but don’t bug
+ us if it doesn’t work with third party clients as you’d hoped.
+
+- multithreading fixes (races, refcounting issues, deadlocks)
+
+- ONVIF audio backchannel fixes
+
+- ONVIF trick mode optimisations
+
+- rtspclientsink: new "update-sdp" signal that allows updating the SDP
+ before sending it to the server via ANNOUNCE. This can be used to
+ add additional metadata to the SDP, for example. The order and
+ number of medias must not be changed, however.
GStreamer VAAPI
-- this section will be filled in in due course
+- new AV1 decoder element (vaapiav1dec)
+
+- H264 decoder: handle stereoscopic 3D video with frame packing
+ arrangement SEI messages
+
+- H265 encoder: added Screen Content Coding extensions support
+
+- H265 decoder: gained MAIN_444_12 profile support (decoded to
+ Y412_LE), and 4:2:2 12-bits support (decoded to Y212_LE)
+
+- vaapipostproc: gained BT2020 color standard support
+
+- vaapidecode: now generates caps templates dynamically at runtime in
+ order to advertise actually supported caps instead of all
+ theoretically supported caps.
+
+- GST_VAAPI_DRM_DEVICE environment variable to force a specified DRM
+ device when a DRM display is used. It is ignored when other types of
+ displays are used. By default /dev/dri/renderD128 is used for DRM
+ display.
GStreamer OMX
-- this section will be filled in in due course
+- subframe support in H.264/H.265 decoders
GStreamer Editing Services and NLE
-- this section will be filled in in due course
+- framepositioner: new "operator" property to access blending modes in
+ the compositor
+- timeline: Implement snapping to markers
+- smart-mixer: Add support for d3d11compositor and glvideomixer
+- titleclip: add "draw-shadow" child property
+- ges:// URI support to define a timeline from a description.
+- command-line-formatter
+ - Add track management to timeline description
+ - Add keyframe support
+- ges-launch-1.0:
+ - Add an interactive mode where we can seek etc…
+ - Add option to forward tags
+ - Allow using an existing clip to determine the rendering format
+ (both topology and profile) via new --profile-from command line
+ argument.
+- Fix static build
GStreamer validate
-- this section will be filled in in due course
+- report: Add a way to force backtraces on reports even if not a
+ critical issue (GST_VALIDATE_ISSUE_FLAGS_FORCE_BACKTRACE)
+- Add a flag to gst_validate_replace_variables_in_string() allow
+ defining how to resolve variables in structs
+- Add gst_validate_bin_monitor_get_scenario() to get the bin monitor
+ scenario, which is useful for applications that use Validate
+ directly.
+- Add an expected-values parameter to wait, message-type=XX allowing
+ more precise filtering of the message we are waiting for.
+- Add config file support: each test can now use a config file for the
+ given media file used to test.
+- Add support to check properties of object properties
+- scenario: Add an "action-done" signal to signal when an action is
+ done
+- scenario: Add a "run-command" action type
+- scenario: Allow forcing running action on idle from scenario file
+- scenario: Allow iterating over arrays in foreach
+- scenario: Rename ‘interlaced’ action to ‘non-blocking’
+- scenario: Add a non-blocking flag to the wait signal
GStreamer Python Bindings
-- this section will be filled in in due course
+- Fixes for Python 3.10
+- Various build fixes
+- at least one known breaking change caused by g-i annotation changes
+ (see below)
GStreamer C# Bindings
-- this section will be filled in in due course
+- Fix GstDebugGraphDetails enum
+- Updated to latests GtkSharp
+- Updated to include GStreamer 1.20 API
GStreamer Rust Bindings and Rust Plugins
-The GStreamer Rust bindings are released separately with a different
-release cadence that’s tied to gtk-rs, but the latest release has
-already been updated for the upcoming new GStreamer 1.20 API.
-
-gst-plugins-rs, the module containing GStreamer plugins written in Rust,
-has also seen lots of activity with many new elements and plugins.
-
-What follows is a list of elements and plugins available in
-gst-plugins-rs, so people don’t miss out on all those potentially useful
-elements that have no C equivalent.
-
-- FIXME: add new elements
-
-Rust audio plugins
-
-- audiornnoise: New element for audio denoising which implements the
- noise removal algorithm of the Xiph RNNoise library, in Rust
-- rsaudioecho: Port of the audioecho element from gst-plugins-good
- rsaudioloudnorm: Live audio loudness normalization element based on
- the FFmpeg af_loudnorm filter
-- claxondec: FLAC lossless audio codec decoder element based on the
- pure-Rust claxon implementation
-- csoundfilter: Audio filter that can use any filter defined via the
- Csound audio programming language
-- lewtondec: Vorbis audio decoder element based on the pure-Rust
- lewton implementation
-
-Rust video plugins
-
-- cdgdec/cdgparse: Decoder and parser for the CD+G video codec based
- on a pure-Rust CD+G implementation, used for example by karaoke CDs
-- cea608overlay: CEA-608 Closed Captions overlay element
-- cea608tott: CEA-608 Closed Captions to timed-text (e.g. VTT or SRT
- subtitles) converter
-- tttocea608: CEA-608 Closed Captions from timed-text converter
-- mccenc/mccparse: MacCaption Closed Caption format encoder and parser
-- sccenc/sccparse: Scenarist Closed Caption format encoder and parser
-- dav1dec: AV1 video decoder based on the dav1d decoder implementation
- by the VLC project
-- rav1enc: AV1 video encoder based on the fast and pure-Rust rav1e
- encoder implementation
-- rsflvdemux: Alternative to the flvdemux FLV demuxer element from
- gst-plugins-good, not feature-equivalent yet
-- rsgifenc/rspngenc: GIF/PNG encoder elements based on the pure-Rust
- implementations by the image-rs project
-
-Rust text plugins
-
-- textwrap: Element for line-wrapping timed text (e.g. subtitles) for
- better screen-fitting, including hyphenation support for some
- languages
-
-Rust network plugins
-
-- reqwesthttpsrc: HTTP(S) source element based on the Rust
- reqwest/hyper HTTP implementations and almost feature-equivalent
- with the main GStreamer HTTP source souphttpsrc
-- s3src/s3sink: Source/sink element for the Amazon S3 cloud storage
-- awstranscriber: Live audio to timed text transcription element using
- the Amazon AWS Transcribe API
-
-Generic Rust plugins
-
-- sodiumencrypter/sodiumdecrypter: Encryption/decryption element based
- on libsodium/NaCl
-- togglerecord: Recording element that allows to pause/resume
- recordings easily and considers keyframe boundaries
-- fallbackswitch/fallbacksrc: Elements for handling potentially
- failing (network) sources, restarting them on errors/timeout and
- showing a fallback stream instead
-- threadshare: Set of elements that provide alternatives for various
- existing GStreamer elements but allow to share the streaming threads
- between each other to reduce the number of threads
-- rsfilesrc/rsfilesink: File source/sink elements as replacements for
- the existing filesrc/filesink elements
+- The GStreamer Rust bindings are released separately with a different
+ release cadence that’s tied to gtk-rs, but the latest release has
+ already been updated for the upcoming new GStreamer 1.20 API (v1_20
+ feature).
+
+- gst-plugins-rs, the module containing GStreamer plugins written in
+ Rust, has also seen lots of activity with many new elements and
+ plugins. See the New Elements section above for a list of new Rust
+ elements.
Build and Dependencies
- Meson 0.59 or newer is required to build GStreamer now.
-- FIXME: this section will be filled in in due course
+- The GLib requirement has been bumped to GLib 2.56 or newer (from
+ March 2018).
+
+- The wpe plugin now requires wpe >= 2.28 and wpebackend-fdo >= 1.8
Explicit opt-in required for build of certain plugins with (A)GPL dependencies
See Building plugins with (A)GPL-licensed dependencies for more details
and a non-exhaustive list of plugins affected.
-gst-build: replaced by Monorepo
+gst-build: replaced by mono repository
-- this section will be filled in in due course
-
-- FIXME: describe + link to Monorepo FAQ
+See mono repository section above and the GStreamer mono repository FAQ.
Cerbero
on platforms where dependencies are not readily available, such as
Windows, Android, iOS and macOS.
-General improvements
+General Cerbero improvements
-- this section will be filled in in due course
+- Plugin removed: libvisual
+- New plugins: rtpmanagerbad and rist
-macOS / iOS
+macOS / iOS specific Cerbero improvements
-- this section will be filled in in due course
+- XCode 12 support
+- macOS OS release support is now future-proof, similar to iOS
+- macOS Apple Silicon (ARM64) cross-compile support has been added
+- macOS Apple Silicon (ARM64) native support is currently experimental
-Windows
+Windows specific Cerbero improvements
-- this section will be filled in in due course
+- Visual Studio 2022 support has been added
+- bootstrap is faster since it requires building fewer build-tools
+ recipes on Windows
+- package is faster due to better scheduling of recipe stages and
+ elimination of unnecessary autotools regeneration
+- The following plugins are no longer built on Windows:
+ - a52dec (another decoder is still available in libav)
+ - dvdread
+ - resindvd
Windows MSI installer
-- this section will be filled in in due course
+- no major changes
-Linux
+Linux specific Cerbero improvements
-- this section will be filled in in due course
+- Fedora, Debian OS release support is now more future-proof
+- Amazon Linux 2 support has been added
-Android
+Android specific Cerbero improvements
-- this section will be filled in in due course
+- no major changes
Platform-specific changes and improvements
Android
-- this section will be filled in in due course
+- No major changes
macOS and iOS
-- this section will be filled in in due course
+- applemedia: add ProRes support to vtenc and vtdec
Windows
-- this section will be filled in in due course
+- On Windows the high-resolution clock is enabled now in the
+ gst-launch-1.0 and gst-play-1.0 command line tools, which provides
+ better clock and timer performance on Windows, at the cost of higher
+ power consumption. By default, without the high-resolution clock
+ enabled, the timer precision on Windows is system-dependent and may
+ be as bad as 15ms which is not good enough for many multimedia
+ applications. Developers may want to do the same in their Windows
+ applications if they think it’s a good idea for their application
+ use case, and depending on the Windows version they target. This is
+ not done automatically by GStreamer because on older Windows
+ versions (pre-Windows 10) this affects a global Windows setting and
+ also there’s a power consumption vs. performance trade-off that may
+ differ from application to application.
+
+- dxgiscreencapsrc now supports resolution changes
+
+- The wasapi2 audio plugin was rewritten and now has a higher rank
+ than the old wasapi plugin since it has a number of additional
+ features such as automatic stream routing, and no
+ known-but-hard-to-fix issues. The plugin is always built if the
+ Windows 10 SDK is available now.
+
+- The wasapi device providers now detect and notify dynamic device
+ additions/removals
+
+- d3d11screencapturesrc: new desktop capture element, including
+ GstDeviceProvider implementation to enumerate/select target monitors
+ for capture.
+
+- Direct3D11/DXVA decoder now supports AV1 and MPEG2 codecs
+ (d3d11av1dec, d3d11mpeg2dec)
+
+- VP9 decoding got more reliable and stable thanks to a newly written
+ codec parser
+
+- Support for decoding interlaced H.264/AVC streams
+
+- Hardware-accelerated video deinterlacing (d3d11deinterlace) and
+ video mixing (d3d11compositor)
+
+- Video mixing with the Direct3D11 API (d3d11compositor)
+
+- MediaFoundation API based hardware encoders gained the ability to
+ receive Direct3D11 textures as an input
+
+- Seungha’s blog post “GStreamer ❤ Windows: A primer on the cool stuff
+ you’ll find in the 1.20 release” describes many of the
+ Windows-related improvements in more detail
Linux
-- this section will be filled in in due course
+- bluez: LDAC Bluetooth audio codec support in a2dpsink and avdtpsink,
+ as well as an LDAC RTP payloader (rtpldacpay) and an LDAC audio
+ encoder (ldacenc)
+
+- kmssink: gained support for NV24, NV61, RGB16/BGR16 formats;
+ auto-detect NVIDIA Tegra driver
Documentation improvements
-- this section will be filled in in due course
+- hardware-accelerated GPU plugins will now no longer always list all
+ the element variants for all available GPUs, since those are
+ system-dependent and it’s confusing for users to see those in the
+ documentation just because the GStreamer developer who generated the
+ docs had multiple GPUs to play with at the time. Instead just show
+ the default elements.
-Possibly Breaking Changes
+Possibly Breaking and Other Noteworthy Behavioural Changes
+
+- gst_parse_launch(), gst_parse_bin_from_description() and friends
+ will now error out when setting properties that don’t exist on
+ top-level bins. They were silently ignored before.
+
+- The GstWebRTC library does not expose any objects anymore with
+ public fields. Instead properties have been added to replace that
+ functionality. If you are accessing such fields in your application,
+ switch to the corresponding properties.
-- FIXME: this section will be filled in in due course
-- MPEG-TS SCTE-35 API changes (FIXME: flesh out)
-- gst_parse_launch() and friends now error out on non-existing
- properties on top-level bins where they would silently fail and
- ignore those before.
- playbin and uridecodebin now emit the source-setup signal before the
element is added to the bin and linked so that the source element is
already configured before any scheduling query comes in, which is
- useful for elements such as appsrc or giostreamsrc. (Merge Request)
+ useful for elements such as appsrc or giostreamsrc.
+
+- The source element inside urisourcebin (used inside uridecodebin3
+ which is used inside playbin3) is no longer called "source". This
+ shouldn’t affect anyone hopefully, because there’s a "setup-source"
+ signal to configure the source element and no one should rely on
+ names of internal elements anyway.
+
+- The vp8enc element now expects bps (bits per second) for the
+ "temporal-scalability-target-bitrate" property, which is consistent
+ with the "target-bitrate" property. Since additional configuration
+ is required with modern libvpx to make temporal scaling work anyway,
+ chances are that very few people will have been using this property
+
+- vp8enc and vp9enc now default to “good quality” for the "deadline"
+ property rather then “best quality”. Having the deadline set to best
+ quality causes the encoder to be absurdly slow, most real-life users
+ will want the good quality tradeoff instead.
+
+- The experimental GstTranscoder library API in gst-plugins-bad was
+ changed from a GObject signal-based notification mechanism to a
+ GstBus/message-based mechanism akin to GstPlayer/GstPlay.
+
+- MPEG-TS SCTE-35 API: semantic change for SCTE-35 splice commands:
+ timestamps passed by the application should be in running time now,
+ since users of the API can’t really be expected to predict the local
+ PTS of the muxer.
+
+- The GstContext used by souphttpsrc to share the session between
+ multiple element instances has changed. Previously it provided
+ direct access to the internal SoupSession object, now it only
+ provides access to an opaque, internal type. This change is
+ necessary because SoupSession is not thread-safe at all and can’t be
+ shared safely between arbitrary external code and souphttpsrc.
+
+- Python bindings: GObject-introspection related Annotation fixes have
+ led to a case of a GstVideo.VideoInfo-related function signature
+ changing in the Python bindings (possibly one or two other cases
+ too). This is for a function that should never have been exposed in
+ the first place though, so the bindings are being updated to throw
+ an exception in that case, and the correct replacement API has been
+ added in form of an override.
Known Issues
-- this section will be filled in in due course
-
-- There are a couple of known WebRTC-related regressions/blockers:
-
- - webrtc: DTLS setup with Chrome is broken
- - webrtcbin: First keyframe is usually lost
+- nothing in particular at this point (but also see possibly breaking
+ changes section above)
Contributors
-- this section will be filled in in due course
+Aaron Boxer, Adam Leppky, Adam Williamson, Alba Mendez, Alejandro
+González, Aleksandr Slobodeniuk, Alexander Vandenbulcke, Alex Ashley,
+Alicia Boya García, Andika Triwidada, Andoni Morales Alastruey, Andrew
+Wesie, Andrey Moiseev, Antonio Ospite, Antonio Rojas, Arthur Crippa
+Búrigo, Arun Raghavan, Ashley Brighthope, Axel Kellermann, Baek, Bastien
+Nocera, Bastien Reboulet, Benjamin Gaignard, Bing Song, Binh Truong,
+Biswapriyo Nath, Brad Hards, Brad Smith, Brady J. Garvin, Branko
+Subasic, Camilo Celis Guzman, Chris Bass, ChrisDuncanAnyvision, Chris
+White, Corentin Damman, Daniel Almeida, Daniel Knobe, Daniel Stone,
+david, David Fernandez, David Keijser, David Phung, Devarsh Thakkar,
+Dinesh Manajipet, Dmitry Samoylov, Dmitry Shusharin, Dominique Martinet,
+Doug Nazar, Ederson de Souza, Edward Hervey, Emmanuel Gil Peyrot,
+Enrique Ocaña González, Ezequiel Garcia, Fabian Orccon, Fabrice
+Fontaine, Fernando Jimenez Moreno, Florian Karydes, Francisco Javier
+Velázquez-García, François Laignel, Frederich Munch, Fredrik Pålsson,
+George Kiagiadakis, Georg Lippitsch, Göran Jönsson, Guido Günther,
+Guillaume Desmottes, Guiqin Zou, Haakon Sporsheim, Haelwenn (lanodan)
+Monnier, Haihao Xiang, Haihua Hu, Havard Graff, He Junyan, Helmut
+Januschka, Henry Wilkes, Hosang Lee, Hou Qi, Ignacio Casal Quinteiro,
+Igor Kovalenko, Ilya Kreymer, Imanol Fernandez, Jacek Tomaszewski, Jade
+Macho, Jakub Adam, Jakub Janků, Jan Alexander Steffens (heftig), Jan
+Schmidt, Jason Carrete, Jason Pereira, Jay Douglass, Jeongki Kim, Jérôme
+Laheurte, Jimmi Holst Christensen, Johan Sternerup, John Hassell, John
+Lindgren, John-Mark Bell, Jonathan Matthew, Jordan Petridis, Jose
+Quaresma, Julian Bouzas, Julien, Kai Uwe Broulik, Kasper Steensig
+Jensen, Kellermann Axel, Kevin Song, Khem Raj, Knut Inge Hvidsten, Knut
+Saastad, Kristofer Björkström, Lars Lundqvist, Lawrence Troup, Lim Siew
+Hoon, Lucas Stach, Ludvig Rappe, Luis Paulo Fernandes de Barros, Luke
+Yelavich, Mads Buvik Sandvei, Marc Leeman, Marco Felsch, Marek Vasut,
+Marian Cichy, Marijn Suijten, Marius Vlad, Markus Ebner, Mart Raudsepp,
+Matej Knopp, Mathieu Duponchelle, Matthew Waters, Matthieu De Beule,
+Mengkejiergeli Ba, Michael de Gans, Michael Olbrich, Michael Tretter,
+Michal Dzik, Miguel Paris, Mikhail Fludkov, mkba, Nazar Mokrynskyi,
+Nicholas Jackson, Nicola Murino, Nicolas Dufresne, Niklas Hambüchen,
+Nikolay Sivov, Nirbheek Chauhan, Olivier Blin, Olivier Crete, Olivier
+Crête, Paul Goulpié, Per Förlin, Peter Boba, P H, Philippe Normand,
+Philipp Zabel, Pieter Willem Jordaan, Piotrek Brzeziński, Rafał
+Dzięgiel, Rafostar, raghavendra, Raghavendra, Raju Babannavar, Raleigh
+Littles III, Randy Li, Randy Li (ayaka), Ratchanan Srirattanamet, Raul
+Tambre, reed.lawrence, Ricky Tang, Robert Rosengren, Robert Swain, Robin
+Burchell, Roman Sivriver, R S Nikhil Krishna, Ruben Gonzalez, Ruslan
+Khamidullin, Sanchayan Maity, Scott Moreau, Sebastian Dröge, Sergei
+Kovalev, Seungha Yang, Sid Sethupathi, sohwan.park, Sonny Piers, Staz M,
+Stefan Brüns, Stéphane Cerveau, Stephan Hesse, Stian Selnes, Stirling
+Westrup, Théo MAILLART, Thibault Saunier, Tim, Timo Wischer, Tim-Philipp
+Müller, Tim Schneider, Tobias Ronge, Tom Schoonjans, Tulio Beloqui,
+tyler-aicradle, U. Artie Eoff, Ung, Val Doroshchuk, VaL Doroshchuk,
+Víctor Manuel Jáquez Leal, Vivek R, Vivia Nikolaidou, Vivienne
+Watermeier, Vladimir Menshakov, Will Miller, Wim Taymans, Xabier
+Rodriguez Calvar, Xavier Claessens, Xℹ Ruoyao, Yacine Bandou, Yinhang
+Liu, youngh.lee, youngsoo.lee, yychao, Zebediah Figura, Zhang yuankun,
+Zhang Yuankun, Zhao, Zhao Zhili, , Aleksandar Topic, Antonio Ospite,
+Bastien Nocera, Benjamin Gaignard, Brad Hards, Carlos Falgueras García,
+Célestin Marot, Corentin Damman, Corentin Noël, Daniel Almeida, Daniel
+Knobe, Danny Smith, Dave Piché, Dmitry Osipenko, Fabrice Fontaine,
+fjmax, Florian Zwoch, Guillaume Desmottes, Haihua Hu, Heinrich Kruger,
+He Junyan, Jakub Adam, James Cowgill, Jan Alexander Steffens (heftig),
+Jean Felder, Jeongki Kim, Jiri Uncovsky, Joe Todd, Jordan Petridis,
+Krystian Wojtas, Marc-André Lureau, Marcin Kolny, Marc Leeman, Mark
+Nauwelaerts, Martin Reboredo, Mathieu Duponchelle, Matthew Waters,
+Mengkejiergeli Ba, Michael Gruner, Nicolas Dufresne, Nirbheek Chauhan,
+Olivier Crête, Philippe Normand, Rafał Dzięgiel, Ralf Sippl, Robert
+Mader, Sanchayan Maity, Sangchul Lee, Sebastian Dröge, Seungha Yang,
+Stéphane Cerveau, Teh Yule Kim, Thibault Saunier, Thomas Klausner, Timo
+Wischer, Tim-Philipp Müller, Tobias Reineke, Tomasz Andrzejak, Trung Do,
+Tyler Compton, Ung, Víctor Manuel Jáquez Leal, Vivia Nikolaidou, Wim
+Taymans, wngecn, Wonchul Lee, wuchang li, Xavier Claessens, Xi Ruoyao,
+Yoshiharu Hirose, Zhao,
… and many others who have contributed bug reports, translations, sent
suggestions or helped testing.
1.20.0
-1.20.0 is scheduled to be released around October/November 2021.
+1.20.0 is scheduled to be released around early February 2022.
Schedule for 1.22
unstable development version leading up to the stable 1.22 release. The
development of 1.21/1.22 will happen in the git main branch.
-The plan for the 1.22 development cycle is yet to be confirmed.
+The plan for the 1.22 development cycle is yet to be confirmed. Assuming
+no major project-wide reorganisations in the 1.22 cycle we might try and
+aim for a release around August 2022.
1.22 will be backwards-compatible to the stable 1.20, 1.18, 1.16, 1.14,
1.12, 1.10, 1.8, 1.6, 1.4, 1.2 and 1.0 release series.
------------------------------------------------------------------------
These release notes have been prepared by Tim-Philipp Müller with
-contributions from …
+contributions from Matthew Waters, Nicolas Dufresne, Nirbheek Chauhan,
+Sebastian Dröge and Seungha Yang.
License: CC BY-SA 4.0
-This is GStreamer gst-rtsp-server 1.19.3.
+This is GStreamer gst-rtsp-server 1.19.90.
GStreamer 1.19 is the development branch leading up to the next major
stable version which will be 1.20.
"construct": false,
"construct-only": false,
"controllable": false,
- "default": "GStreamer/1.19.3.1",
+ "default": "GStreamer/1.19.90",
"mutable": "null",
"readable": true,
"type": "gchararray",
<release>
<Version>
+ <revision>1.19.90</revision>
+ <branch>main</branch>
+ <name></name>
+ <created>2022-01-28</created>
+ <file-release rdf:resource="https://gstreamer.freedesktop.org/src/gst-rtsp-server/gst-rtsp-server-1.19.90.tar.xz" />
+ </Version>
+ </release>
+
+ <release>
+ <Version>
<revision>1.19.3</revision>
<branch>main</branch>
<name></name>
project('gst-rtsp-server', 'c',
- version : '1.19.3.1',
+ version : '1.19.90',
meson_version : '>= 0.59',
default_options : ['warning_level=1', 'buildtype=debugoptimized'])
-project('gstreamer-sharp', ['cs', 'c'], version: '1.19.3.1',
+project('gstreamer-sharp', ['cs', 'c'], version: '1.19.90',
meson_version : '>= 0.59', license: 'LGPL')
if host_machine.system() == 'osx'
public const string ENCODING_CATEGORY_ONLINE_SERVICE = @"online-service";
public const string ENCODING_CATEGORY_STORAGE_EDITING = @"storage-editing";
public const int PLUGINS_BASE_VERSION_MAJOR = 1;
- public const int PLUGINS_BASE_VERSION_MICRO = 3;
+ public const int PLUGINS_BASE_VERSION_MICRO = 90;
public const int PLUGINS_BASE_VERSION_MINOR = 19;
- public const int PLUGINS_BASE_VERSION_NANO = 1;
+ public const int PLUGINS_BASE_VERSION_NANO = 0;
#endregion
}
}
public const int VALUE_LESS_THAN = -1;
public const int VALUE_UNORDERED = 2;
public const int VERSION_MAJOR = 1;
- public const int VERSION_MICRO = 3;
+ public const int VERSION_MICRO = 90;
public const int VERSION_MINOR = 19;
- public const int VERSION_NANO = 1;
+ public const int VERSION_NANO = 0;
#endregion
}
}
<constant value="1" ctype="gint" gtype="gint" name="VALUE_GREATER_THAN" />
<constant value="-1" ctype="gint" gtype="gint" name="VALUE_LESS_THAN" />
<constant value="2" ctype="gint" gtype="gint" name="VALUE_UNORDERED" />
- <constant value="1" ctype="gint" gtype="gint" name="VERSION_MAJOR" />
- <constant value="3" ctype="gint" gtype="gint" name="VERSION_MICRO" />
- <constant value="19" ctype="gint" gtype="gint" name="VERSION_MINOR" />
- <constant value="1" ctype="gint" gtype="gint" name="VERSION_NANO" />
+ <constant value="1" ctype="gint" gtype="gint" name="VERSION_MAJOR" />
+ <constant value="90" ctype="gint" gtype="gint" name="VERSION_MICRO" />
+ <constant value="19" ctype="gint" gtype="gint" name="VERSION_MINOR" />
+ <constant value="0" ctype="gint" gtype="gint" name="VERSION_NANO" />
</object>
<class name="Parse" cname="GstParse" disable_void_ctor="1">
<method name="ParseBinFromDescription" cname="gst_parse_bin_from_description" shared="true">
<constant value="file-extension" ctype="gchar*" gtype="gchar*" name="ENCODING_CATEGORY_FILE_EXTENSION" />
<constant value="online-service" ctype="gchar*" gtype="gchar*" name="ENCODING_CATEGORY_ONLINE_SERVICE" />
<constant value="storage-editing" ctype="gchar*" gtype="gchar*" name="ENCODING_CATEGORY_STORAGE_EDITING" />
- <constant value="1" ctype="gint" gtype="gint" name="PLUGINS_BASE_VERSION_MAJOR" />
- <constant value="3" ctype="gint" gtype="gint" name="PLUGINS_BASE_VERSION_MICRO" />
- <constant value="19" ctype="gint" gtype="gint" name="PLUGINS_BASE_VERSION_MINOR" />
- <constant value="1" ctype="gint" gtype="gint" name="PLUGINS_BASE_VERSION_NANO" />
+ <constant value="1" ctype="gint" gtype="gint" name="PLUGINS_BASE_VERSION_MAJOR" />
+ <constant value="90" ctype="gint" gtype="gint" name="PLUGINS_BASE_VERSION_MICRO" />
+ <constant value="19" ctype="gint" gtype="gint" name="PLUGINS_BASE_VERSION_MINOR" />
+ <constant value="0" ctype="gint" gtype="gint" name="PLUGINS_BASE_VERSION_NANO" />
</object>
</namespace>
<namespace name="Gst.Rtp" library="gstrtp-1.0-0.dll">
+=== release 1.19.90 ===
+
+2022-01-28 14:28:35 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * NEWS:
+ * RELEASE:
+ * gstreamer-vaapi.doap:
+ * meson.build:
+ Release 1.19.90
+
+2022-01-28 14:28:28 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * ChangeLog:
+ Update ChangeLogs for 1.19.90
+
2022-01-05 02:07:59 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
* docs/meson.build:
GStreamer 1.20 Release Notes
GStreamer 1.20 has not been released yet. It is scheduled for release in
-November 2021.
+late January / early February 2022.
1.19.x is the unstable development version that is being developed in
-the git main branch and which will eventually result in 1.20, and 1.19.3
-is the current development release in that series
-
-It is expected that feature freeze will be in early October 2021,
-followed by one or two 1.19.9x pre-releases and the new 1.20 stable
-release around October/November 2021.
+the git main branch and which will eventually result in 1.20, and
+1.19.90 is the first release candidate in that series (1.20rc1).
1.20 will be backwards-compatible to the stable 1.18, 1.16, 1.14, 1.12,
-1.10, 1.8, 1.6,, 1.4, 1.2 and 1.0 release series.
+1.10, 1.8, 1.6, 1.4, 1.2 and 1.0 release series.
See https://gstreamer.freedesktop.org/releases/1.20/ for the latest
version of this document.
-Last updated: Monday 1 November 2021, 01:00 UTC (log)
+Last updated: Wednesday 26 January 2022, 01:00 UTC (log)
Introduction
Highlights
-- this section will be completed in due course
+- Development in GitLab was switched to a single git repository
+ containing all the modules
+- GstPlay: new high-level playback library, replaces GstPlayer
+- WebM Alpha decoding support
+- Encoding profiles can now be tweaked with additional
+ application-specified element properties
+- Compositor: multi-threaded video conversion and mixing
+- RTP header extensions: unified support in RTP depayloader and
+ payloader base classes
+- SMPTE 2022-1 2-D Forward Error Correction support
+- Smart encoding (passthrough) support for VP8, VP9, H.265 in
+ encodebin and transcodebin
+- Runtime compatibility support for libsoup2 and libsoup3 (libsoup3
+ support experimental)
+- Video decoder subframe support
+- Video decoder automatic packet-loss, data corruption, and keyframe
+ request handling for RTP / WebRTC / RTSP
+- MP4 and Matroska muxers now support profile/level/resolution changes
+ for H264/H265 input streams (i.e. codec data changing on the fly)
+- MP4 muxing mode that initially creates a fragmented mp4 which is
+ converted to a regular mp4 on EOS
+- Audio support for the WebKit Port for Embedded (WPE) web page source
+ element
+- CUDA based video color space convert and rescale elements and
+ upload/download elements
+- NVIDIA memory:NVMM support for OpenGL glupload and gldownload
+ elements
+- Many WebRTC improvements
+- The new VA-API plugin implemention fleshed out with more decoders
+ and new postproc elements
+- AppSink API to retrieve events in addition to buffers and buffer
+ lists
+- AppSrc gained more configuration options for the internal queue
+ (leakiness, limits in buffers and time, getters to read current
+ levels)
+- Updated Rust bindings and many new Rust plugins
+- Improved support for custom minimal GStreamer builds
+- Support build against FFmpeg 5.0
+- Linux Stateless CODEC support gained MPEG2 and VP9
+- Windows Direct3D11/DXVA decoder gained AV1 and MPEG2 support
+- Lots of new plugins, features, performance improvements and bug
+ fixes
Major new features and changes
Noteworthy new features and API
-- this section will be filled in in due course
+- gst_element_get_request_pad() has been deprecated in favour of the
+ newly-added gst_element_request_pad_simple() which does the exact
+ same thing but has a less confusing name that hopefully makes clear
+ that the function request a new pad rather than just retrieves an
+ already-existing request pad.
+
+Development in GitLab was switched to a single git repository containing all the modules
+
+The GStreamer multimedia framework is a set of libraries and plugins
+split into a number of distinct modules which are released independently
+and which have so far been developed in separate git repositories in
+freedesktop.org GitLab.
+
+In addition to these separate git repositories there was a gst-build
+module that would use the Meson build systems’s subproject feature to
+download each individual module and then build everything in one go. It
+would also provide an uninstalled development environment that made it
+easy to work on GStreamer and use or test versions other than the
+system-installed GStreamer version.
+
+All of these modules have now (as of 28 September 2021) been merged into
+a single git repository (“Mono repository” or “monorepo”) which should
+simplify development workflows and continuous integration, especially
+where changes need to be made to multiple modules at once.
+
+This mono repository merge will primarily affect GStreamer developers
+and contributors and anyone who has workflows based on the GStreamer git
+repositories.
+
+The Rust bindings and Rust plugins modules have not been merged into the
+mono repository at this time because they follow a different release
+cycle.
+
+The mono repository lives in the existing GStreamer core git repository
+in GitLab in the new main branch and all future development will happen
+on this branch.
+
+Modules will continue to be released as separate tarballs.
+
+For more details, please see the GStreamer mono repository FAQ.
+
+GstPlay: new high-level playback library replacing GstPlayer
+
+- GstPlay is a new high-level playback library that replaces the older
+ GstPlayer API. It is basically the same API as GstPlayer but
+ refactored to use bus messages for application notifications instead
+ of GObject signals. There is still a signal adapter object for those
+ who prefer signals. Since the existing GstPlayer API is already in
+ use in various applications, it didn’t seem like a good idea to
+ break it entirely. Instead a new API was added, and it is expected
+ that this new GstPlay API will be moved to gst-plugins-base in
+ future.
+
+- The existing GstPlayer API is scheduled for deprecation and will be
+ removed at some point in the future (e.g. in GStreamer 1.24), so
+ application developers are urged to migrate to the new GstPlay API
+ at their earliest convenience.
+
+WebM alpha decoding
+
+- Implement WebM alpha decoding (VP8/VP9 with alpha), which required
+ support and additions in various places. This is supported both with
+ software decoders and hardware-accelerated decoders.
+
+- VP8/VP9 don’t support alpha components natively in the codec, so the
+ way this is implemented in WebM is by encoding the alpha plane with
+ transparency data as a separate VP8/VP9 stream. Inside the WebM
+ container (a variant of Matroska) this is coded as a single video
+ track with the “normal” VP8/VP9 video data making up the main video
+ data and each frame of video having an encoded alpha frame attached
+ to it as extra data ("BlockAdditional").
+
+- matroskademux has been extended extract this per-frame alpha side
+ data and attach it in form of a GstVideoCodecAlphaMeta to the
+ regular video buffers. Note that this new meta is specific to this
+ VP8/VP9 alpha support and can’t be used to just add alpha support to
+ other codecs that don’t support it. Lastly, matroskademux also
+ advertises the fact that the streams contain alpha in the caps.
+
+- The new codecalpha plugin contains various bits of infrastructure to
+ support autoplugging and debugging:
+
+ - codecalphademux splits out the alpha stream from the metas on
+ the regular VP8/VP9 buffers
+ - alphacombine takes two decoded raw video streams (one alpha, one
+ the regular video) and combines it into a video stream with
+ alpha
+ - vp8alphadecodebin + vp9alphadecodebin are wrapper bins that use
+ the regular vp8dec and vp9dec software decoders to decode
+ regular and alpha streams and combine them again. To decodebin
+ these look like regular decoders which ju
+ - The V4L2 CODEC plugin has stateless VP8/VP9 decoders that can
+ decode both alpha and non-alpha stream with a single decoder
+ instance
+
+- A new AV12 video format was added which is basically NV12 with an
+ alpha plane, which is more convenient for many hardware-accelerated
+ decoders.
+
+- Watch Nicolas Dufresne’s LCA 2022 talk “Bringing WebM Alpha support
+ to GStreamer” for all the details and a demo.
+
+RTP Header Extensions Base Class and Automatic Header Extension Handling in RTP Payloaders and Depayloaders
+
+- RTP Header Extensions are specified in RFC 5285 and provide a way to
+ add small pieces of data to RTP packets in between the RTP header
+ and the RTP payload. This is often used for per-frame metadata,
+ extended timestamps or other application-specific extra data. There
+ are several commonly-used extensions specified in various RFCs, but
+ senders are free to put any kind of data in there, as long as sender
+ and receiver both know what that data is. Receivers that don’t know
+ about the header extensions will just skip the extra data without
+ ever looking at it. These header extensions can often be combined
+ with any kind of payload format, so may need to be supported by many
+ RTP payloader and depayloader elements.
+
+- Inserting and extracting RTP header extension data has so far been a
+ bit inconvenient in GStreamer: There are functions to add and
+ retrieve RTP header extension data from RTP packets, but nothing
+ works automatically, even for common extensions. People would have
+ to do the insertion/extraction either in custom elements
+ before/after the RTP payloader/depayloader, or inside pad probes,
+ which isn’t very nice.
+
+- This release adds various pieces of new infrastructure for generic
+ RTP header extension handling, as well as some implementations for
+ common extensions:
+
+ - GstRTPHeaderExtension is a new helper base class for reading and
+ writing RTP header extensions. Nominally this subclasses
+ GstElement, but only so these extensions are stored in the
+ registry where they can be looked up by URI or name. They don’t
+ have pads and don’t get added to the pipeline graph as an
+ element.
+
+ - "add-extension" and "clear-extension" action signals on RTP
+ payloaders and depayloaders for manual extension management
+
+ - The "request-extension" signal will be emitted if an extension
+ is encountered that requires explicit mapping by the application
+
+ - new "auto-header-extension" property on RTP payloaders and
+ depayloaders for automatic handling of known header extensions.
+ This is enabled by default. The extensions must be signalled via
+ caps / SDP.
+
+ - RTP header extension implementations:
+
+ - rtphdrextclientaudiolevel: Client-to-Mixer Audio Level
+ Indication (RFC 6464) (also see below)
+ - rtphdrextcolorspace: Color Space extension, extends RTP
+ packets with color space and high dynamic range (HDR)
+ information
+ - rtphdrexttwcc: Transport Wide Congestion Control support
+
+- gst_rtp_buffer_remove_extension_data() is a new helper function to
+ remove an RTP header extension from an RTP buffer
+
+- The existing gst_rtp_buffer_set_extension_data() now also supports
+ shrinking the extension data in size
+
+AppSink and AppSrc improvements
+
+- appsink: new API to pull events out of appsink in addition to
+ buffers and buffer lists.
+
+ There was previously no way for users to receive incoming events
+ from appsink properly serialised with the data flow, even if they
+ are serialised events. The reason for that is that the only way to
+ intercept events was via a pad probe on the appsink sink pad, but
+ there is also internal queuing inside of appsink, so it’s difficult
+ to ascertain the right order of everything in all cases.
+
+ There is now a new "new-serialized-event" signal which will be
+ emitted when there’s a new event pending (just like the existing
+ "new-sample" signal). The "emit-signals" property must be set to
+ TRUE in order to activate this (but it’s also fine to just pull from
+ the application thread without using the signals).
+
+ gst_app_sink_pull_object() and gst_app_sink_try_pull_object() can be
+ used to pull out either an event or a new sample carrying a buffer
+ or buffer list, whatever is next in the queue.
+
+ EOS events will be filtered and will not be returned. EOS handling
+ can be done the ususal way, same as with _pull_sample().
+
+- appsrc: allow configuration of internal queue limits in time and
+ buffers and add leaky mode.
+
+ There is internal queuing inside appsrc so the application thread
+ can push data into the element which will then be picked up by the
+ source element’s streaming thread and pushed into the pipeline from
+ that streaming thread. This queue is unlimited by default and until
+ now it was only possible to set a maximum size limit in bytes. When
+ that byte limit is reached, the pushing thread (application thread)
+ would be blocked until more space becomes available.
+
+ A limit in bytes is not particularly useful for many use cases, so
+ now it is possible to also configure limits in time and buffers
+ using the new "max-time" and "max-buffers" properties. Of course
+ there are also matching new read-only"current-level-buffers" and
+ "current-level-time properties" properties to query the current fill
+ level of the internal queue in time and buffers.
+
+ And as if that wasn’t enough the internal queue can also be
+ configured as leaky using the new "leaky-type" property. That way
+ when the queue is full the application thread won’t be blocked when
+ it tries to push in more data, but instead either the new buffer
+ will be dropped or the oldest data in the queue will be dropped.
+
+Better string serialization of nested GstCaps and GstStructures
+
+- New string serialisation format for structs and caps that can handle
+ nested structs and caps properly by using brackets to delimit nested
+ items (e.g. some-struct, some-field=[nested-struct, nested=true]).
+ Unlike the default format the new variant can also support more than
+ one level of nesting. For backwards-compatibility reasons the old
+ format is still output by default when serialising caps and structs
+ using the existing API. The new functions gst_caps_serialize() and
+ gst_structure_serialize() can be used to output strings in the new
+ format.
+
+Convenience API for custom GstMetas
+
+- New convenience API to register and create custom GstMetas:
+ gst_meta_register_custom() and gst_buffer_add_custom_meta(). Such
+ custom meta is backed by a GstStructure and does not require that
+ users of the API expose their GstMeta implementation as public API
+ for other components to make use of it. In addition, it provides a
+ simpler interface by ignoring the impl vs. api distinction that the
+ regular API exposes. This new API is meant to be the meta
+ counterpart to custom events and messages, and to be more convenient
+ than the lower-level API when the absolute best performance isn’t a
+ requirement. The reason it’s less performant than a “proper” meta is
+ that a proper meta is just a C struct in the end whereas this goes
+ through the GstStructure API which has a bit more overhead, which
+ for most scenarios is negligible however. This new API is useful for
+ experimentation or proprietary metas, but also has some limitations:
+ it can only be used if there’s a single producer of these metas;
+ it’s not allowed to register the same custom meta multiple times or
+ from multiple places.
+
+Additional Element Properties on Encoding Profiles
+
+- GstEncodingProfile: The new "element-properties" and
+ gst_encoding_profile_set_element_properties() API allows
+ applications to set additional element properties on encoding
+ profiles to configure muxers and encoders. So far the encoding
+ profile template was the only place where this could be specified,
+ but often what applications want to do is take a ready-made encoding
+ profile shipped by GStreamer or the application and then tweak the
+ settings on top of that, which is now possible with this API. Since
+ applications can’t always know in advance what encoder element will
+ be used in the end, it’s even possible to specify properties on a
+ per-element basis.
+
+ Encoding Profiles are used in the encodebin, transcodebin and
+ camerabin elements and APIs to configure output formats (containers
+ and elementary streams).
+
+Audio Level Indication Meta for RFC 6464
+
+- New GstAudioLevelMeta containing Audio Level Indication as per RFC
+ 6464
+
+- The level element has been updated to add GstAudioLevelMeta on
+ buffers if the "audio-level-meta" property is set to TRUE. This can
+ then in turn be picked up by RTP payloaders to signal the audio
+ level to receivers through RTP header extensions (see above).
+
+- New Client-to-Mixer Audio Level Indication (RFC6464) RTP Header
+ Extension which should be automatically created and used by RTP
+ payloaders and depayloaders if their "auto-header-extension"
+ property is enabled and if the extension is part of the RTP caps.
+
+Automatic packet loss, data corruption and keyframe request handling for video decoders
+
+- The GstVideoDecoder base class has gained various new APIs to
+ automatically handle packet loss and data corruption better by
+ default, especially in RTP, RTSP and WebRTC streaming scenarios, and
+ to give subclasses more control about how they want to handle
+ missing data:
+
+ - Video decoder subclasses can mark output frames as corrupted via
+ the new GST_VIDEO_CODEC_FRAME_FLAG_CORRUPTED flag
+
+ - A new "discard-corrupted-frames" property allows applications to
+ configure decoders so that corrupted frames are directly
+ discarded instead of being forwarded inside the pipeline. This
+ is a replacement for the "output-corrupt" property of the FFmpeg
+ decoders.
+
+ - RTP depayloaders can now signal to decoders that data is missing
+ when sending GAP events for lost packets. GAP events can be sent
+ for various reason in a GStreamer pipeline. Often they are just
+ used to let downstream elements know that there isn’t a buffer
+ available at the moment, so downstream elements can move on
+ instead of waiting for one. They are also sent by RTP
+ depayloaders in the case that packets are missing, however, and
+ so far a decoder was not able to differentiate the two cases.
+ This has been remedied now: GAP events can be decorated with
+ gst_event_set_gap_flags() and GST_GAP_FLAG_MISSING_DATA to let
+ decoders now what happened, and decoders can then use that in
+ some cases to handle missing data better.
+
+ - The GstVideoDecoder::handle_missing_data vfunc was added to
+ inform subclasses about packet loss or missing data and let them
+ handle it in their own way if they like.
+
+ - gst_video_decoder_set_needs_sync_point() lets subclasses signal
+ that they need the stream to start with a sync point. If
+ enabled, the base class will discard all non-sync point frames
+ in the beginning and after a flush and does not pass them to the
+ subclass. Furthermore, if the first frame is not a sync point,
+ the base class will try and request a sync frame from upstream
+ by sending a force-key-unit event (see next items).
+
+ - New "automatic-request-sync-points" and
+ "automatic-request-sync-point-flags" properties to automatically
+ request sync points when needed, e.g. on packet loss or if the
+ first frame is not a keyframe. Applications may want to enable
+ this on decoders operating in e.g. RTP/WebRTC/RTSP receiver
+ pipelines.
+
+ - The new "min-force-key-unit-interval" property can be used to
+ ensure there’s a minimal interval between keyframe requests to
+ upstream (and/or the sender) and we’re not flooding the sender
+ with key unit requests.
+
+ - gst_video_decoder_request_sync_point() allows subclasses to
+ request a new sync point (e.g. if they choose to do their own
+ missing data handling). This will still honour the
+ "min-force-key-unit-interval" property if set.
+
+Improved support for custom minimal GStreamer builds
+
+- Element registration and registration of other plugin features
+ inside plugin init functions has been improved in order to
+ facilitate minimal custom GStreamer builds.
+
+- A number of new macros have been added to declare and create
+ per-element and per-pluginfeature register functions in all plugins,
+ and then call those from the per-plugin plugin_init functions:
+
+ - GST_ELEMENT_REGISTER_DEFINE,
+ GST_DEVICE_PROVIDER_REGISTER_DEFINE,
+ GST_DYNAMIC_TYPE_REGISTER_DEFINE, GST_TYPE_FIND_REGISTER_DEFINE
+ for the actual registration call with GStreamer
+ - GST_ELEMENT_REGISTER, GST_DEVICE_PROVIDER_REGISTER,
+ GST_DYNAMIC_TYPE_REGISTER, GST_PLUGIN_STATIC_REGISTER,
+ GST_TYPE_FIND_REGISTER to call the registration function defined
+ by the REGISTER_DEFINE macro
+ - GST_ELEMENT_REGISTER_DECLARE,
+ GST_DEVICE_PROVIDER_REGISTER_DECLARE,
+ GST_DYNAMIC_TYPE_REGISTER_DECLARE,
+ GST_TYPE_FIND_REGISTER_DECLARE to declare the registration
+ function defined by the REGISTER_DEFINE macro
+ - and various variants for advanced use cases.
+
+- This means that applications can call the per-element and
+ per-pluginfeature registration functions for only the elements they
+ need instead of registering plugins as a whole with all kinds of
+ elements that may not be required (e.g. encoder and decoder instead
+ of just decoder). In case of static linking all unused functions and
+ their dependencies would be removed in this case by the linker,
+ which helps minimise binary size for custom builds.
+
+- gst_init() will automatically call a gst_init_static_plugins()
+ function if one exists.
+
+- See the GStreamer static build documentation and Stéphane’s blog
+ post Generate a minimal GStreamer build, tailored to your needs for
+ more details.
New elements
-- this section will be filled in in due course
+- New aesdec and aesenc elements for AES encryption and decryption in
+ a custom format.
+
+- New encodebin2 element with dynamic/sometimes source pads in order
+ to support the option of doing the muxing outside of encodebin,
+ e.g. in combination with a splitmuxsink.
+
+- New fakeaudiosink and videocodectestsink elements for testing and
+ debugging (see below for more details)
+
+- rtpisacpay, rtpisacdepay: new RTP payloader and depayloader for iSAC
+ audio codec
+
+- rtpst2022-1-fecdec, rtpst2022-1-fecenc: new elements providing SMPTE
+ 2022-1 2-D Forward Error Correction. More details in Mathieu’s blog
+ post.
+
+- isac: new plugin wrapping the Internet Speech Audio Codec reference
+ encoder and decoder from the WebRTC project.
+
+- asio: plugin for Steinberg ASIO (Audio Streaming Input/Output) API
+
+- gssrc, gssink: add source and sink for Google Cloud Storage
+
+- onnx: new plugin to apply ONNX neural network models to video
+
+- openaptx: aptX and aptX-HD codecs using libopenaptx (v0.2.0)
+
+- qroverlay, debugqroverlay: new elements that allows overlaying data
+ on top of video in form of a QR code
+
+- cvtracker: new OpenCV-based tracker element
+
+- av1parse, vp9parse: new parsers for AV1 and VP9 video
+
+- va: work on the new VA-API plugin implementation for
+ hardware-accelerated video decoding and encoding has continued at
+ pace, with various new decoders and filters having joined the
+ initial vah264dec:
+
+ - vah265dec: VA-API H.265 decoder
+ - vavp8dec: VA-API VP8 decoder
+ - vavp9dec: VA-API VP9 decoder
+ - vaav1dec: VA-API AV1 decoder
+ - vampeg2dec: VA-API MPEG-2 decoder
+ - vadeinterlace: : VA-API deinterlace filter
+ - vapostproc: : VA-API postproc filter (color conversion,
+ resizing, cropping, color balance, video rotation, skin tone
+ enhancement, denoise, sharpen)
+
+ See Víctor’s blog post “GstVA in GStreamer 1.20” for more details
+ and what’s coming up next.
+
+- vaapiav1dec: new AV1 decoder element (in gstreamer-vaapi)
+
+- msdkav1dec: hardware-accelerated AV1 decoder using the Intel Media
+ SDK / oneVPL
+
+- nvcodec plugin for NVIDIA NVCODEC API for hardware-accelerated video
+ encoding and decoding:
+
+ - cudaconvert, cudascale: new CUDA based video color space convert
+ and rescale elements
+ - cudaupload, cudadownload: new helper elements for memory
+ transfer between CUDA and system memory spaces
+ - nvvp8sldec, nvvp9sldec: new GstCodecs-based VP8/VP9 decoders
+
+- Various new hardware-accelerated elements for Windows:
+
+ - d3d11screencapturesrc: new desktop capture element, including a
+ GstDeviceProvider implementation to enumerate/select target
+ monitors for capture.
+ - d3d11av1dec and d3d11mpeg2dec: AV1 and MPEG-2 decoders
+ - d3d11deinterlace: deinterlacing filter
+ - d3d11compositor: video composing element
+ - see Windows section below for more details
+
+- new Rust plugins:
+
+ - audiornnoise: Removes noise from an audio stream
+ - awstranscribeparse: Parses AWS audio transcripts into timed text
+ buffers
+ - ccdetect: Detects if valid closed captions are present in a
+ closed captions stream
+ - cea608tojson: Converts CEA-608 Closed Captions to a JSON
+ representation
+ - cmafmux: CMAF fragmented MP4 muxer
+ - dashmp4mux: DASH fragmented MP4 muxer
+ - isofmp4mux: ISO fragmented MP4 muxer
+ - ebur128level: EBU R128 Loudness Level Measurement
+ - ffv1dec: FFV1 video decoder
+ - gtk4paintablesink: GTK4 video sink, which provides a
+ GdkPaintable that can be rendered in various widgets
+ - hlssink3: HTTP Live Streaming sink
+ - hrtfrender: Head-Related Transfer Function (HRTF) renderer
+ - hsvdetector: HSV colorspace detector
+ - hsvfilter: HSV colorspace filter
+ - jsongstenc: Wraps buffers containing any valid top-level JSON
+ structures into higher level JSON objects, and outputs those as
+ ndjson
+ - jsongstparse: Parses ndjson as output by jsongstenc
+ - jsontovtt: converts JSON to WebVTT subtitles
+ - regex: Applies regular expression operations on text
+ - roundedcorners: Adds rounded corners to video
+ - spotifyaudiosrc: Spotify source
+ - textahead: Display upcoming text buffers ahead (e.g. for
+ Karaoke)
+ - transcriberbin: passthrough bin that transcribes raw audio to
+ closed captions using awstranscriber and puts the captions as
+ metas onto the video
+ - tttojson: Converts timed text to a JSON representation
+ - uriplaylistbin: Playlist source bin
+ - webpdec-rs: WebP image decoder with animation support
+
+- New plugin codecalpha with elements to assist with WebM Alpha
+ decoding
+
+ - codecalphademux: Split stream with GstVideoCodecAlphaMeta into
+ two streams
+ - alphacombine: Combine two raw video stream (I420 or NV12) as one
+ stream with alpha channel (A420 or AV12)
+ - vp8alphadecodebin: A bin to handle software decoding of VP8 with
+ alpha
+ - vp9alphadecodebin: A bin to handle software decoding of VP9 with
+ alpha
+
+- New hardware accelerated elements for Linux:
+
+ - v4l2slmpeg2dec: Support for Linux Stateless MPEG2 decoders
+ - v4l2slvp9dec: Support for Linux Stateless VP9 decoders
+ - v4l2slvp8alphadecodebin: Support HW accelerated VP8 with alpha
+ layer decoding
+ - v4l2slvp9alphadecodebin: Support HW accelerated VP9 with alpha
+ layer decoding
New element features and additions
-- this section will be filled in in due course
+- assrender: handle more font mime types; better interaction with
+ matroskademux for embedded fonts
+
+- audiobuffersplit: Add support for specifying output buffer size in
+ bytes (not just duration)
+
+- audiolatency: new "samplesperbuffer" property so users can configure
+ the number of samples per buffer. The default value is 240 samples
+ which is equivalent to 5ms latency with a sample rate of 48000,
+ which might be larger than actual buffer size of audio capture
+ device.
+
+- audiomixer, audiointerleave, GstAudioAggregator: now keep a count of
+ samples that are dropped or processed as statistic and can be made
+ to post QoS messages on the bus whenever samples are dropped by
+ setting the "qos-messages" property on input pads.
+
+- audiomixer, compositor: improved handling of new inputs added at
+ runtime. New API was added to the GstAggregator base class to allow
+ subclasses to opt into an aggregation mode where inactive pads are
+ ignored when processing input buffers
+ (gst_aggregator_set_ignore_inactive_pads(),
+ gst_aggregator_pad_is_inactive()). An “inactive pad” in this context
+ is a pad which, in live mode, hasn’t yet received a first buffer,
+ but has been waited on at least once. What would happen usually in
+ this case is that the aggregator would wait for data on this pad
+ every time, up to the maximum configured latency. This would
+ inadvertently push mixer elements in live mode to the configured
+ latency envelope and delay processing when new inputs are added at
+ runtime until these inputs have actually produced data. This is
+ usually undesirable. With this new API, new inputs can be added
+ (requested) and configured and they won’t delay the data processing.
+ Applications can opt into this new behaviour by setting the
+ "ignore-inactive-pads" property on compositor, audiomixer or other
+ GstAudioAggregator-based elements.
+
+- cccombiner: implement “scheduling” of captions. So far cccombiner’s
+ behaviour was essentially that of a funnel: it strictly looked at
+ input timestamps to associate together video and caption buffers.
+ Now it will try to smoothly schedule caption buffers in order to
+ have exactly one per output video buffer. This might involve
+ rewriting input captions, for example when the input is CDP then
+ sequence counters are rewritten, time codes are dropped and
+ potentially re-injected if the input video frame had a time code
+ meta. This can also lead to the input drifting from synchronization,
+ when there isn’t enough padding in the input stream to catch up. In
+ that case the element will start dropping old caption buffers once
+ the number of buffers in its internal queue reaches a certain limit
+ (configurable via the "max-scheduled" property). The new original
+ funnel-like behaviour can be restored by setting the "scheduling"
+ property to FALSE.
+
+- ccconverter: new "cdp-mode" property to specify which sections to
+ include in CDP packets (timecode, CC data, service info). Various
+ software, including ffmpeg’s Decklink support, fails parsing CDP
+ packets that contain anything but CC data in the CDP packets.
+
+- clocksync: new "sync-to-first" property for automatic timestamp
+ offset setup: if set clocksync will set up the "ts-offset" value
+ based on the first buffer and the pipeline’s running time when the
+ first buffer arrived. The newly configured "ts-offset" in this case
+ would be the value that allows outputting the first buffer without
+ waiting on the clock. This is useful for example to feed a non-live
+ input into an already-running pipeline.
+
+- compositor:
+
+ - multi-threaded input conversion and compositing. Set the
+ "max-threads" property to activate this.
+ - new "sizing-policy" property to support display aspect ratio
+ (DAR)-aware scaling. By default the image is scaled to fill the
+ configured destination rectangle without padding and without
+ keeping the aspect ratio. With sizing-policy=keep-aspect-ratio
+ the input image is scaled to fit the destination rectangle
+ specified by GstCompositorPad:{xpos, ypos, width, height}
+ properties preserving the aspect ratio. As a result, the image
+ will be centered in the destination rectangle with padding if
+ necessary.
+ - new "zero-size-is-unscaled" property on input pads. By default
+ pad width=0 or pad height=0 mean that the stream should not be
+ scaled in that dimension. But if the "zero-size-is-unscaled"
+ property is set to FALSE a width or height of 0 is instead
+ interpreted to mean that the input image on that pad should not
+ be composited, which is useful when creating animations where an
+ input image is made smaller and smaller until it disappears.
+ - improved handling of new inputs at runtime via
+ "ignore-inactive-pads"property (see above for details)
+ - allow output format with alpha even if none of the inputs have
+ alpha (also glvideomixer and other GstVideoAggregator
+ subclasses)
+
+- dashsink: add h265 codec support and signals for allowing custom
+ playlist/fragment output
+
+- decodebin3:
+
+ - improved decoder selection, especially for hardware decoders
+ - make input activation “atomic” when adding inputs dynamically
+ - better interleave handling: take into account decoder latency
+ for interleave size
+
+- decklink:
+
+ - Updated DeckLink SDK to 11.2 to support DeckLink 8K Pro
+ - decklinkvideosrc:
+ - More accurate and stable capture timestamps: use the
+ hardware reference clock time when the frame was finished
+ being captured instead of a clock time much further down the
+ road.
+ - Automatically detect widescreen vs. normal NTSC/PAL
+
+- encodebin:
+
+ - add “smart encoding” support for H.265, VP8 and VP9 (i.e. only
+ re-encode where needed and otherwise pass through encoded video
+ as-is).
+ - H264/H265 smart encoding improvements: respect user-specified
+ stream-format, but if not specified default to avc3/hvc1 with
+ in-band SPS/PPS/VPS signalling for more flexibility.
+ - new encodebin2 element with dynamic/sometimes source pads in
+ order to support the option of doing the muxing outside of
+ encodebin, e.g. in combination with splitmuxsink.
+ - add APIs to set element properties on encoding profiles (see
+ below)
+
+- errorignore: new "ignore-eos" property to also ignore FLOW_EOS from
+ downstream elements
+
+- giosrc: add support for growing source files: applications can
+ specify that the underlying file being read is growing by setting
+ the "is-growing" property. If set, the source won’t EOS when it
+ reaches the end of the file, but will instead start monitoring it
+ and will start reading data again whenever a change is detected. The
+ new "waiting-data" and "done-waiting-data" signals keep the
+ application informed about the current state.
+
+- gtksink, gtkglsink:
+
+ - scroll event support: forwarded as navigation events into the
+ pipeline
+ - "video-aspect-ratio-override" property to force a specific
+ aspect ratio
+ - "rotate-method" property and support automatic rotation based on
+ image tags
+
+- identity: new "stats" property allows applications to retrieve the
+ number of bytes and buffers that have passed through so far.
+
+- interlace: add support for more formats, esp 10-bit, 12-bit and
+ 16-bit ones
+
+- jack: new "low-latency" property for automatic latency-optimized
+ setting and "port-names" property to select ports explicitly
+
+- jpegdec: support output conversion to RGB using libjpeg-turbo (for
+ certain input files)
+
+- line21dec:
+
+ - "mode" property to control whether and how detected closed
+ captions should be inserted in the list of existing close
+ caption metas on the input frame (if any): add, drop, or
+ replace.
+ - "ntsc-only" property to only look for captions if video has NTSC
+ resolution
+
+- line21enc: new "remove-caption-meta" to remove metas from output
+ buffers after encoding the captions into the video data; support for
+ CDP closed captions
+
+- matroskademux, matroskamux: Add support for ffv1, a lossless
+ intra-frame video coding format.
+
+- matroskamux: accept in-band SPS/PPS/VPS for H264 and H265
+ (i.e. stream-format avc3 and hev1) which allows on-the-fly
+ profile/level/resolution changes.
+
+- matroskamux: new "cluster-timestamp-offset" property, useful for use
+ cases where the container timestamps should map to some absolute
+ wall clock time, for example.
+
+- rtpsrc: add "caps" property to allow explicit setting of the caps
+ where needed
+
+- mpegts: support SCTE-35 passthrough via new "send-scte35-events"
+ property on MPEG-TS demuxer tsdemux. When enabled, SCTE 35 sections
+ (eg ad placement opportunities) are forwarded as events donwstream
+ where they can be picked up again by mpegtsmux. This required a
+ semantic change in the SCTE-35 section API: timestamps are now in
+ running time instead of muxer pts.
+
+- tsdemux: Handle PCR-less MPEG-TS streams; more robust timestamp
+ handling in certain corner cases and for poorly muxed streams.
+
+- mpegtsmux:
+
+ - More conformance improvements to make MPEG-TS analyzers happy:
+ - PCR timing accuracy: Improvements to the way mpegtsmux
+ outputs PCR observations in CBR mode, so that a PCR
+ observation is always inserted when needed, so that we never
+ miss the configured pcr-interval, as that triggers various
+ MPEG-TS analyser errors.
+ - Improved PCR/SI scheduling
+ - Don’t write PCR until PAT/PMT are output to make sure streams
+ start cleanly with a PAT/PMT.
+ - Allow overriding the automatic PMT PID selection via
+ application-supplied PMT_%d fields in the prog-map
+ structure/property.
+
+- mp4mux:
+
+ - new "first-moov-then-finalise" mode for fragmented output where
+ the output will start with a self-contained moov atom for the
+ first fragment, and then produce regular fragments. Then at the
+ end when the file is finalised, the initial moov is invalidated
+ and a new moov is written covering the entire file. This way the
+ file is a “fragmented mp4” file while it is still being written
+ out, and remains playable at all times, but at the end it is
+ turned into a regular mp4 file (with former fragment headers
+ remaining as unused junk data in the file).
+ - support H.264 avc3 and H.265 hvc1 stream formats as input where
+ the codec data is signalled in-band inside the bitstream instead
+ of caps/file headers.
+ - support profile/level/resolution changes for H264/H265 input
+ streams (i.e. codec data changing on the fly). Each codec_data
+ is put into its own SampleTableEntry inside the stsd, unless the
+ input is in avc3 stream format in which case it’s written
+ in-band an not in the headers.
+
+- multifilesink: new ""min-keyframe-distance"" property to make
+ minimum distance between keyframes in next-file=key-frame mode
+ configurable instead of hard-coding it to 10 seconds.
+
+- mxfdemux has seen a big refactoring to support non-frame wrappings
+ and more accurate timestamp/seek handling for some formats
+
+- msdk plugin for hardware-accelerated video encoding and decoding
+ using the Intel Media SDK:
+
+ - oneVPL support (Intel oneAPI Video Processing Library)
+ - AV1 decoding support
+ - H.264 decoder now supports constrained-high and progressive-high
+ profiles
+ - H.264 encoder:
+ - more configuration options (properties):
+ "intra-refresh-type", "min-qp" , "max-qp", "p-pyramid",
+ "dblk-idc"
+ - H.265 encoder:
+ - can output main-still-picture profile
+ - now inserts HDR SEIs (mastering display colour volume and
+ content light level)
+ - more configuration options (properties):
+ "intra-refresh-type", "min-qp" , "max-qp", "p-pyramid",
+ "b-pyramid", "dblk-idc", "transform-skip"
+ - support for RGB 10bit format
+ - External bitrate control in encoders
+ - Video post proc element msdkvpp gained support for 12-bit pixel
+ formats P012_LE, Y212_LE and Y412_LE
+
+- nvh264sldec: interlaced stream support
+
+- openh264enc: support main, high, constrained-high and
+ progressive-high profiles
+
+- openjpeg: support for multithreaded decoding and encoding
+
+- rtspsrc: now supports IPv6 also for tunneled mode (RTSP-over-HTTP);
+ new "ignore-x-server-reply" property to ignore the
+ x-server-ip-address server header reply in case of HTTP tunneling,
+ as it is often broken.
+
+- souphttpsrc: Runtime compatibility support for libsoup2 and
+ libsoup3. libsoup3 is the latest major version of libsoup, but
+ libsoup2 and libsoup3 can’t co-exist in the same process because
+ there is no namespacing or versioning for GObject types. As a
+ result, it would be awkward if the GStreamer souphttpsrc plugin
+ linked to a specific version of libsoup, because it would only work
+ with applications that use the same version of libsoup. To make this
+ work, the soup plugin now tries to determine the libsoup version
+ used by the application (and its other dependencies) at runtime on
+ systems where GStreamer is linked dynamically. libsoup3 support is
+ still considered somewhat experimental at this point.
+
+- srtsrc, srtsink: add signals for the application to accept/reject
+ incoming connections
+
+- timeoverlay: new elapsed-running-time time mode which shows the
+ running time since the first running time (and each flush-stop).
+
+- udpsrc: new timestamping mode to retrieve packet receive timestamps
+ from the kernel via socket control messages (SO_TIMESTAMPNS) on
+ supported platforms
+
+- uritranscodebin: new setup-source and element-setup signals for
+ applications to configure elements used
+
+- v4l2codecs plugin gained support for 4x4 and 32x32 tile formats
+ enabling some platforms or direct renders. Important memory usage
+ improvement.
+
+- v4l2slh264dec now implements the final Linux uAPI as shipped on
+ Linux 5.11 and later.
+
+- valve: add "drop-mode" property and provide two new modes of
+ operation: in drop-mode=forward-sticky-events sticky events
+ (stream-start, segment, tags, caps, etc.) are forwarded downstream
+ even when dropping is enabled; drop-mode=transform-to-gap will in
+ addition also convert buffers into gap events when dropping is
+ enabled, which lets downstream elements know that time is advancing
+ and might allow for preroll in many scenarios. By default all events
+ and all buffers are dropped when dropping is enabled, which can
+ cause problems with caps negotiation not progressing or branches not
+ prerolling when dropping is enabled.
+
+- videocrop: support for many more pixel formats, e.g. planar YUV
+ formats with > 8bits and GBR* video formats; can now also accept
+ video not backed by system memory as long as downstream supports the
+ GstCropMeta
+
+- videotestsrc: new smpte-rp-219 pattern for SMPTE75 RP-219 conformant
+ color bars
+
+- vp8enc: finish support for temporal scalability: two new properties
+ ("temporal-scalability-layer-flags",
+ "temporal-scalability-layer-sync-flags") and a unit change on the
+ "temporal-scalability-target-bitrate" property (now expects bps);
+ also make temporal scalability details available to RTP payloaders
+ as buffer metadata.
+
+- vp9enc: new properties to tweak encoder performance:
+
+ - "aq-mode" to configure adaptive quantization modes
+ - "frame-parallel-decoding" to configure whether to create a
+ bitstream that reduces decoding dependencies between frames
+ which allows staged parallel processing of more than one video
+ frames in the decoder. (Defaults to TRUE)
+ - "row-mt", "tile-columns" and "tile-rows" so multithreading can
+ be enabled on a per-tile basis, instead of on a per tile-column
+ basis. In combination with the new "tile-rows" property, this
+ allows the encoder to make much better use of the available CPU
+ power.
+
+- vp9dec, vp9enc: add support for 10-bit 4:2:0 and 4:2:2 YUV, as well
+ as 8-bit 4:4:4
+
+- vp8enc, vp9enc now default to “good quality” for the deadline
+ property rather then “best quality”. Having the deadline set to best
+ quality causes the encoder to be absurdly slow, most real-life users
+ will prefer good-enough quality with better performance instead.
+
+- wpesrc:
+
+ - implement audio support: a new sometimes source pad will be
+ created for each audio stream created by the web engine.
+ - move wpesrc to wpevideosrc and add a wrapper bin wpesrc to also
+ support audio
+ - also handles web:// URIs now (same as cefsrc)
+ - post messages with the estimated load progress on the bus
-Plugin and library moves
+- x265enc: add negative DTS support, which means timestamps are now
+ offset by 1h same as with x264enc
+
+RTP Payloaders and Depayloaders
+
+- rtpisacpay, rtpisacdepay: new RTP payloader and depayloader for iSAC
+ audio codec
+
+- rtph264depay:
+
+ - new "request-keyframe" property to make the depayloader
+ automatically request a new keyframe from the sender on packet
+ loss, consistent with the new property on rtpvp8depay.
+ - new "wait-for-keyframe" property to make depayloader wait for a
+ new keyframe at the beginning and after packet loss (only
+ effective if the depayloader outputs AUs), consistent with the
+ existing property on rtpvp8depay.
+
+- rtpopuspay, rtpopusdepay: support libwebrtc-compatible multichannel
+ audio in addition to the previously supported multichannel audio
+ modes
+
+- rtpopuspay: add DTX (Discontinuous Transmission) support
+
+- rtpvp8depay: new "request-keyframe" property to make the depayloader
+ automatically request a new keyframe from the sender on packet loss.
+
+- rtpvp8pay: temporal scaling support
-- this section will be filled in in due course
+- rtpvp9depay: Improved SVC handling (aggregate all layers)
+
+RTP Infrastructure
+
+- rtpst2022-1-fecdec, rtpst2022-1-fecenc: new elements providing SMPTE
+ 2022-1 2-D Forward Error Correction. More details in Mathieu’s blog
+ post.
+
+- rtpreddec: BUNDLE support
+
+- rtpredenc, rtpulpfecenc: add support for Transport-wide Congestion
+ Control (TWCC)
+
+- rtpsession: new "twcc-feedback-interval" property to allow RTCP TWCC
+ reports to be scheduled on a timer instead of per marker-bit.
+
+Plugin and library moves
- There were no plugin moves or library moves in this cycle.
The following elements or plugins have been removed:
-- this section will be filled in in due course
-
- The ofa audio fingerprinting plugin has been removed. The MusicIP
database has been defunct for years so this plugin is likely neither
useful nor used by anyone.
Miscellaneous API additions
-- this section will be filled in in due course
+Core
+
+- gst_buffer_new_memdup() is a convenience function for the
+ widely-used gst_buffer_new_wrapped(g_memdup(data,size),size)
+ pattern.
+
+- gst_caps_features_new_single() creates a new single GstCapsFeatures,
+ avoiding the need to use the vararg function with NULL terminator
+ for simple cases.
+
+- gst_element_type_set_skip_documentation() can be used by plugins to
+ signal that certain elements should not be included in the GStreamer
+ plugin documentation. This is useful for plugins where elements are
+ registered dynamically based on hardware capabilities and/or where
+ the available plugins and properties vary from system to system.
+ This is used in the d3d11 plugin for example to ensure that only the
+ list of default elements is advertised in the documentation.
+
+- gst_type_find_suggest_empty_simple() is a new convenience function
+ for typefinders for cases where there’s only a media type and no
+ other fields.
+
+- New API to create elements and set properties at construction time,
+ which is not only convenient, but also allows GStreamer elements to
+ have construct-only properties: gst_element_factory_make_full(),
+ gst_element_factory_make_valist(),
+ gst_element_factory_make_with_properties(),
+ gst_element_factory_create_full(),
+ gst_element_factory_create_valist(),
+ gst_element_factory_create_with_properties().
+
+- GstSharedTaskPool: new “shared” task pool subclass with slightly
+ different default behaviour than the existing GstTaskPool which
+ would create unlimited number of threads for new tasks. The shared
+ taskpool creates up to N threads (default: 1) and then distributes
+ pending tasks to those threads round-robin style, and blocks if no
+ thread is available. It is possible to join tasks. This can be used
+ by plugins to implement simple multi-threaded processing and is used
+ for the new multi-threaded video conversion and compositing done in
+ GstVideoAggregator, videoconverter and compositor.
+
+Plugins Base Utils library
+
+- GstDiscoverer:
+
+ - gst_discoverer_container_info_get_tags() was added to retrieve
+ global/container tags (vs. per-stream tags). Per-Stream tags can
+ be retrieved via the existing
+ gst_discoverer_stream_info_get_tags().
+ gst_discoverer_info_get_tags(), which for many files returns a
+ confusing mix of stream and container tags, has been deprecated
+ in favour of the container/stream-specific functions.
+ - gst_discoverer_stream_info_get_stream_number() returns a unique
+ integer identifier for a given stream within the given
+ GstDiscoverer context. (If this matches the stream number inside
+ the container bitstream that’s by coincidence and not by
+ design.)
+
+- gst_pb_utils_get_caps_description_flags() can be used to query
+ whether certain caps represent a container, audio, video, image,
+ subtitles, tags, or something else. This only works for formats
+ known to GStreamer.
+
+- gst_pb_utils_get_file_extension_from_caps() returns a possible file
+ extension for given caps.
+
+- gst_codec_utils_h264_get_profile_flags_level(): Parses profile,
+ flags, and level from H264 AvcC codec_data. The format of H264 AVCC
+ extradata/sequence_header is documented in the ITU-T H.264
+ specification section 7.3.2.1.1 as well as in ISO/IEC 14496-15
+ section 5.3.3.1.2.
+
+- gst_codec_utils_caps_get_mime_codec() to convert caps to a RFC 6381
+ compatible MIME codec string codec. Useful for providing the codecs
+ field inside the Content-Type HTTP header for containerized formats,
+ such as mp4 or matroska.
+
+GStreamer OpenGL integration library and plugins
+
+- glcolorconvert: added suppport for converting the video formats
+ A420, AV12, BGR, BGRA, RGBP and BGRP.
+
+- Added support to GstGLBuffer for persistent buffer mappings where a
+ Pixel Buffer Object (PBO) can be mapped by both the CPU and the GPU.
+ This removes a memcpy() when uploading textures or vertices
+ particularly when software decoders (e.g. libav) are direct
+ rendering into our memory. Improves transfer performance
+ significantly. Requires OpenGL 4.4, GL_ARB_buffer_storage or
+ GL_EXT_buffer_storage
+
+- Added various helper functions for handling 4x4 matrices of affine
+ transformations as used by GstVideoAffineTransformationMeta.
+
+- Add support to GstGLContext for allowing the application to control
+ the config (EGLConfig, GLXConfig, etc) used when creating the OpenGL
+ context. This allows the ability to choose between RGB16 or RGB10A2
+ or RGBA8 back/front buffer configurations that were previously
+ hardcoded. GstGLContext also supports retrieving the configuration
+ it was created with or from an externally provide OpenGL context
+ handle. This infrastructure is also used to create a compatible
+ config from an application/externally provided OpenGL context in
+ order to improve compatibility with other OpenGL frameworks and GUI
+ toolkits. A new environment variable GST_GL_CONFIG was also added to
+ be able to request a specific configuration from the command line.
+ Note: different platforms will have different functionality
+ available.
+
+- Add support for choosing between EGL and WGL at runtime when running
+ on Windows. Previously this was a build-time switch. Allows use in
+ e.g. Gtk applications on Windows that target EGL/ANGLE without
+ recompiling GStreamer. gst_gl_display_new_with_type() can be used by
+ applications to choose a specific display type to use.
+
+- Build fixes to explicitly check for Broadcom-specific libraries on
+ older versions of the Raspberry Pi platform. The Broadcom OpenGL ES
+ and EGL libraries have different filenames. Using the vc4 Mesa
+ driver on the Raspberry Pi is not affected.
+
+- Added support to glupload and gldownload for transferring RGBA
+ buffers using the memory:NVMM available on the Nvidia Tegra family
+ of embedded devices.
+
+- Added support for choosing libOpenGL and libGLX as used in a GLVND
+ environment on unix-based platforms. This allows using desktop
+ OpenGL and EGL without pulling in any GLX symbols as would be
+ required with libGL.
+
+Video library
+
+- New raw video formats:
+
+ - AV12 (NV12 with alpha plane)
+ - RGBP and BGRP (planar RGB formats)
+ - ARGB64 variants with specified endianness instead of host
+ endianness:
+ - ARGB64_LE, ARGB64_BE
+ - RGBA64_BE, RGBA64_LE
+ - BGRA64_BE, BGRA64_LE
+ - ABGR64_BE, ABGR64_LE
+
+- gst_video_orientation_from_tag() is new convenience API to parse the
+ image orientation from a GstTagList.
+
+- GstVideoDecoder subframe support (see below)
+
+- GstVideoCodecState now also carries some HDR metadata
+
+- Ancillary video data: implement transform functions for AFD/Bar
+ metas, so they will be forwarded in more cases
+
+MPEG-TS library
+
+This library only handles section parsing and such, see above for
+changes to the actual mpegtsmux and mpegtsdemux elements.
+
+- many additions and improvements to SCTE-35 section parsing
+- new API for fetching extended descriptors:
+ gst_mpegts_find_descriptor_with_extension()
+- add support for SIT sections (Selection Information Tables)
+- expose event-from-section constructor gst_event_new_mpegts_section()
+- parse Audio Preselection Descriptor needed for Dolby AC-4
+
+GstWebRTC library + webrtcbin
+
+- Change the way in which sink pads and transceivers are matched
+ together to support easier usage. If a pad is created without a
+ specific index (i.e. using sink_%u as the pad template), then an
+ available compatible transceiver will be searched for. If a specific
+ index is requested (i.e. sink_1) then if a transceiver for that
+ m-line already exists, that transceiver must match the new sink pad
+ request. If there is no transceiver available in either scenario, a
+ new transceiver is created. If a mixture of both sink_1 and sink_%u
+ requests result in an impossible situation, an error will be
+ produced at pad request time or from create offer/answer.
+
+- webrtcbin now uses regular ICE nomination instead of libnice’s
+ default of aggressive ICE nomination. Regular ICE nomination is the
+ default recommended by various relevant standards and improves
+ connectivity in specific network scenarios.
+
+- Add support for limiting the port range used for RTP with the
+ addition of the min-rtp-port and max-rtp-port properties on the ICE
+ object.
+
+- Expose the SCTP transport as a property on webrtcbin to more closely
+ match the WebRTC specification.
+
+- Added support for taking into account the data channel transport
+ state when determining the value of the "connection-state" property.
+ Previous versions of the WebRTC spec did not include the data
+ channel state when computing this value.
+
+- Add configuration for choosing the size of the underlying sockets
+ used for transporting media data
+
+- Always advertise support for the transport-cc RTCP feedback protocol
+ as rtpbin supports it. For full support, the configured caps (input
+ or through codec-preferences) need to include the relevant RTP
+ header extension.
+
+- Numerous fixes to caps and media handling to fail-fast when an
+ incompatible situation is detected.
+
+- Improved support for attaching the required media after a remote
+ offer has been set.
+
+- Add support for dynamically changing the amount of FEC used for a
+ particular stream.
+
+- webrtcbin now stops further SDP processing at the first error it
+ encounters.
+
+- Completed support for either local or the remote closing a data
+ channel.
+
+- Various fixes when performing BUNDLEing of the media streams in
+ relation to RTX and FEC usage.
+
+- Add support for writing out QoS DSCP marking on outgoing packets to
+ improve reliability in some network scenarios.
+
+- Improvements to the statistics returned by the get-stats signal
+ including the addition of the raw statistics from the internal
+ RTPSource, the TWCC stats when available.
+
+- The webrtc library does not expose any objects anymore with public
+ fields. Instead properties have been added to replace that
+ functionality. If you are accessing such fields in your application,
+ switch to the corresponding properties.
+
+GstCodecs and Video Parsers
+
+- Support for render delays to improve throughput across all CODECs
+ (used with NVDEC and V4L2).
+- lots of improvements to parsers and the codec parsing decoder base
+ classes (H264, H265, VP8, VP9, AV1, MPEG-2) used for various
+ hardware-accelerate decoder APIs.
+
+Bindings support
+
+- gst_allocation_params_new() allocates a GstAllocationParams struct
+ on the heap. This should only be used by bindings (and freed via
+ gst_allocation_params_free() then). In C code you would allocate
+ this on the stack and only init it in place.
+
+- gst_debug_log_literal() can be used to log a string to the debug log
+ without going through any printf format expansion and associated
+ overhead. This is mostly useful for bindings such as the Rust
+ bindings which may have done their own formatting already .
+
+- Provide non-inlined versions of refcounting APIs for various
+ GStreamer mini objects, so that they can be consumed by bindings
+ (e.g. gstreamer-sharp): gst_buffer_ref, gst_buffer_unref,
+ gst_clear_buffer, gst_buffer_copy, gst_buffer_replace,
+ gst_buffer_list_ref, gst_buffer_list_unref, gst_clear_buffer_list,
+ gst_buffer_list_copy, gst_buffer_list_replace, gst_buffer_list_take,
+ gst_caps_ref, gst_caps_unref, gst_clear_caps, gst_caps_replace,
+ gst_caps_take, gst_context_ref, gst_context_unref, gst_context_copy,
+ gst_context_replace, gst_event_replace, gst_event_steal,
+ gst_event_take, gst_event_ref, gst_event_unref, gst_clear_event,
+ gst_event_copy, gst_memory_ref, gst_memory_unref, gst_message_ref,
+ gst_message_unref, gst_clear_message, gst_message_copy,
+ gst_message_replace, gst_message_take, gst_promise_ref,
+ gst_promise_unref, gst_query_ref, gst_query_unref, gst_clear_query,
+ gst_query_copy, gst_query_replace, gst_query_take, gst_sample_ref,
+ gst_sample_unref, gst_sample_copy, gst_tag_list_ref,
+ gst_tag_list_unref, gst_clear_tag_list, gst_tag_list_replace,
+ gst_tag_list_take, gst_uri_copy, gst_uri_ref, gst_uri_unref,
+ gst_clear_uri.
+
+- expose a GType for GstMiniObject
+
+- gst_device_provider_probe() now returns non-floating device object
+
+API Deprecations
+
+- gst_element_get_request_pad() has been deprecated in favour of the
+ newly-added gst_element_request_pad_simple() which does the exact
+ same thing but has a less confusing name that hopefully makes clear
+ that the function request a new pad rather than just retrieves an
+ already-existing request pad.
+
+- gst_discoverer_info_get_tags(), which for many files returns a
+ confusing mix of stream and container tags, has been deprecated in
+ favour of the container-specific and stream-specific functions,
+ gst_discoverer_container_info_get_tags() and
+ gst_discoverer_stream_info_get_tags().
+
+- gst_video_sink_center_rect() was deprecated in favour of the more
+ generic newly-added gst_video_center_rect().
+
+- The GST_MEMORY_FLAG_NO_SHARE flag has been deprecated, as it tends
+ to cause problems and prevents sub-buffering. If pooling or lifetime
+ tracking is required, memories should be allocated through a custom
+ GstAllocator instead of relying on the lifetime of the buffers the
+ memories were originally attached to, which is fragile anyway.
+
+- The GstPlayer high-level playback library is being replaced with the
+ new GstPlay library (see above). GstPlayer should be considered
+ deprecated at this point and will be marked as such in the next
+ development cycle. Applications should be ported to GstPlay.
+
+- Gstreamer Editing Services: ges_video_transition_set_border(),
+ ges_video_transition_get_border()
+ ges_video_transition_set_inverted()
+ ges_video_transition_is_inverted() have been deprecated, use
+ ges_timeline_element_set_children_properties() instead.
Miscellaneous performance, latency and memory optimisations
-- this section will be filled in in due course
+More video conversion fast paths
-Miscellaneous other changes and enhancements
+- v210 ↔ I420, YV12, Y42B, UYVY and YUY2
+- A420 → RGB
+
+Less jitter when waiting on the system clock
+
+- Better system clock wait accuracy, less jitter: where available,
+ clock_nanosleep is used for higher accuracy for waits below 500
+ usecs, and waits below 2ms will first use the regular waiting system
+ and then clock_nanosleep for the remainder. The various wait
+ implementation have a latency ranging from 50 to 500+ microseconds.
+ While this is not a major issue when dealing with a low number of
+ waits per second (for ex: video), it does introduce a non-negligible
+ jitter for synchronisation of higher packet rate systems.
+
+Video decoder subframe support
-- this section will be filled in in due course
+- The GstVideoDecoder base class gained API to process input at the
+ sub-frame level. That way video decoders can start decoding slices
+ before they have received the full input frame in its entirety (to
+ the extent this is supported by the codec, of course). This helps
+ with CPU utilisation and reduces latency.
-Tracing framework and debugging improvements
+- This functionality is now being used in the OpenJPEG JPEG 2000
+ decoder, the FFmpeg H.264 decoder (in case of NAL-aligned input) and
+ the OpenMAX H.264/H.265 decoders (in case of NAL-aligned input).
-- this section will be filled in in due course
+Miscellaneous other changes and enhancements
+
+- GstDeviceMonitor no longer fails to start just because one of the
+ device providers failed to start. That could happen for example on
+ systems where the pulseaudio device provider is installed, but
+ pulseaudio isn’t actually running but ALSA is used for audio
+ instead. In the same vein the device monitor now keeps track of
+ which providers have been started (via the new
+ gst_device_provider_is_started()) and only stops actually running
+ device providers when stopping the device monitor.
+
+- On embedded systems it can be useful to create a registry that can
+ be shared and read by multiple processes running as different users.
+ It is now possible to set the new GST_REGISTRY_MODE environment
+ variable to specify the file mode for the registry file, which by
+ default is set to be only user readable/writable.
+
+- GstNetClientClock will signal lost sync in case the remote time
+ resets (e.g. because device power cycles), by emitting the “synced”
+ signal with synced=FALSE parameter, so applications can take action.
+
+- gst_value_deserialize_with_pspec() allows deserialization with a
+ hint for what the target GType should be. This allows for example
+ passing arrays of flags through the command line or
+ gst_util_set_object_arg(), eg: foo="<bar,bar+baz>".
+
+- It’s now allowed to create an empty GstVideoOverlayComposition
+ without any rectangles by passing a NULL rectangle to
+ gst_video_overlay_composition_new(). This is useful for bindings and
+ simplifies application code in some places.
+
+Tracing framework, debugging and testing improvements
+
+- New factories tracer to list loaded elements (and other plugin
+ features). This can be useful to collect a list of elements needed
+ for an application, which then in turn can be used to create a
+ tailored minimal GStreamer build that contains just the elements
+ needed and nothing else.
+- New plugin-feature-loaded tracing hook for use by tracers like the
+ new factories tracer
+
+- GstHarness: Add gst_harness_set_live() so that harnesses can be set
+ to non-live and return is-live=false in latency queries if needed.
+ Default behaviour is to always return is-live=true in latency
+ queries.
+
+- navseek: new "hold-eos" property. When enabled, the element will
+ hold back an EOS event until the next keystroke (via navigation
+ events). This can be used to keep a video sink showing the last
+ frame of a video pipeline until a key is pressed instead of tearing
+ it down immediately on EOS.
+
+- New fakeaudiosink element: mimics an audio sink and can be used for
+ testing and CI pipelines on systems where no audio system is
+ installed or running. It differs from fakesink in that it only
+ support audio caps and syncs to the clock by default like a normal
+ audio sink. It also implements the GstStreamVolume interface like
+ most audio sinks do.
+
+- New videocodectestsink element for video codec conformance testing:
+ Calculates MD5 checksums for video frames and skips any padding
+ whilst doing so. Can optionally also write back the video data with
+ padding removed into a file for easy byte-by-byte comparison with
+ reference data.
Tools
-- this section will be filled in in due course
+gst-inspect-1.0
+
+- Can sort the list of plugins by passing --sort=name as command line
+ option
+
+gst-launch-1.0
+
+- will now error out on top-level properties that don’t exist and
+ which were silently ignored before
+- On Windows the high-resolution clock is enabled now, which provides
+ better clock and timer performance on Windows (see Windows section
+ below for more details).
+
+gst-play-1.0
+
+- New --start-position command line argument to start playback from
+ the specified position
+- Audio can be muted/unmuted in interactive mode by pressing the m
+ key.
+- On Windows the high-resolution clock is enabled now (see Windows
+ section below for more details)
+
+gst-device-monitor-1.0
+
+- New --include-hidden command line argument to also show “hidden”
+ device providers
+
+ges-launch-1.0
+
+- New interactive mode that allows seeking and such. Can be disabled
+ by passing the --no-interactive argument on the command line.
+- Option to forward tags
+- Allow using an existing clip to determine the rendering format (both
+ topology and profile) via new --profile-from command line argument.
GStreamer RTSP server
-- this section will be filled in in due course
+- GstRTSPMediaFactory gained API to disable RTCP
+ (gst_rtsp_media_factory_set_enable_rtcp(), "enable-rtcp" property).
+ Previously RTCP was always allowed for all RTSP medias. With this
+ change it is possible to disable RTCP completely, no matter if the
+ client wants to do RTCP or not.
+
+- Make a mount point of / work correctly. While not allowed by the
+ RTSP 2 spec, the RTSP 1 spec is silent on this and it is used in the
+ wild. It is now possible to use / as a mount path in
+ gst-rtsp-server, e.g. rtsp://example.com/ would work with this now.
+ Note that query/fragment parts of the URI are not necessarily
+ correctly handled, and behaviour will differ between various
+ client/server implementations; so use it if you must but don’t bug
+ us if it doesn’t work with third party clients as you’d hoped.
+
+- multithreading fixes (races, refcounting issues, deadlocks)
+
+- ONVIF audio backchannel fixes
+
+- ONVIF trick mode optimisations
+
+- rtspclientsink: new "update-sdp" signal that allows updating the SDP
+ before sending it to the server via ANNOUNCE. This can be used to
+ add additional metadata to the SDP, for example. The order and
+ number of medias must not be changed, however.
GStreamer VAAPI
-- this section will be filled in in due course
+- new AV1 decoder element (vaapiav1dec)
+
+- H264 decoder: handle stereoscopic 3D video with frame packing
+ arrangement SEI messages
+
+- H265 encoder: added Screen Content Coding extensions support
+
+- H265 decoder: gained MAIN_444_12 profile support (decoded to
+ Y412_LE), and 4:2:2 12-bits support (decoded to Y212_LE)
+
+- vaapipostproc: gained BT2020 color standard support
+
+- vaapidecode: now generates caps templates dynamically at runtime in
+ order to advertise actually supported caps instead of all
+ theoretically supported caps.
+
+- GST_VAAPI_DRM_DEVICE environment variable to force a specified DRM
+ device when a DRM display is used. It is ignored when other types of
+ displays are used. By default /dev/dri/renderD128 is used for DRM
+ display.
GStreamer OMX
-- this section will be filled in in due course
+- subframe support in H.264/H.265 decoders
GStreamer Editing Services and NLE
-- this section will be filled in in due course
+- framepositioner: new "operator" property to access blending modes in
+ the compositor
+- timeline: Implement snapping to markers
+- smart-mixer: Add support for d3d11compositor and glvideomixer
+- titleclip: add "draw-shadow" child property
+- ges:// URI support to define a timeline from a description.
+- command-line-formatter
+ - Add track management to timeline description
+ - Add keyframe support
+- ges-launch-1.0:
+ - Add an interactive mode where we can seek etc…
+ - Add option to forward tags
+ - Allow using an existing clip to determine the rendering format
+ (both topology and profile) via new --profile-from command line
+ argument.
+- Fix static build
GStreamer validate
-- this section will be filled in in due course
+- report: Add a way to force backtraces on reports even if not a
+ critical issue (GST_VALIDATE_ISSUE_FLAGS_FORCE_BACKTRACE)
+- Add a flag to gst_validate_replace_variables_in_string() allow
+ defining how to resolve variables in structs
+- Add gst_validate_bin_monitor_get_scenario() to get the bin monitor
+ scenario, which is useful for applications that use Validate
+ directly.
+- Add an expected-values parameter to wait, message-type=XX allowing
+ more precise filtering of the message we are waiting for.
+- Add config file support: each test can now use a config file for the
+ given media file used to test.
+- Add support to check properties of object properties
+- scenario: Add an "action-done" signal to signal when an action is
+ done
+- scenario: Add a "run-command" action type
+- scenario: Allow forcing running action on idle from scenario file
+- scenario: Allow iterating over arrays in foreach
+- scenario: Rename ‘interlaced’ action to ‘non-blocking’
+- scenario: Add a non-blocking flag to the wait signal
GStreamer Python Bindings
-- this section will be filled in in due course
+- Fixes for Python 3.10
+- Various build fixes
+- at least one known breaking change caused by g-i annotation changes
+ (see below)
GStreamer C# Bindings
-- this section will be filled in in due course
+- Fix GstDebugGraphDetails enum
+- Updated to latests GtkSharp
+- Updated to include GStreamer 1.20 API
GStreamer Rust Bindings and Rust Plugins
-The GStreamer Rust bindings are released separately with a different
-release cadence that’s tied to gtk-rs, but the latest release has
-already been updated for the upcoming new GStreamer 1.20 API.
-
-gst-plugins-rs, the module containing GStreamer plugins written in Rust,
-has also seen lots of activity with many new elements and plugins.
-
-What follows is a list of elements and plugins available in
-gst-plugins-rs, so people don’t miss out on all those potentially useful
-elements that have no C equivalent.
-
-- FIXME: add new elements
-
-Rust audio plugins
-
-- audiornnoise: New element for audio denoising which implements the
- noise removal algorithm of the Xiph RNNoise library, in Rust
-- rsaudioecho: Port of the audioecho element from gst-plugins-good
- rsaudioloudnorm: Live audio loudness normalization element based on
- the FFmpeg af_loudnorm filter
-- claxondec: FLAC lossless audio codec decoder element based on the
- pure-Rust claxon implementation
-- csoundfilter: Audio filter that can use any filter defined via the
- Csound audio programming language
-- lewtondec: Vorbis audio decoder element based on the pure-Rust
- lewton implementation
-
-Rust video plugins
-
-- cdgdec/cdgparse: Decoder and parser for the CD+G video codec based
- on a pure-Rust CD+G implementation, used for example by karaoke CDs
-- cea608overlay: CEA-608 Closed Captions overlay element
-- cea608tott: CEA-608 Closed Captions to timed-text (e.g. VTT or SRT
- subtitles) converter
-- tttocea608: CEA-608 Closed Captions from timed-text converter
-- mccenc/mccparse: MacCaption Closed Caption format encoder and parser
-- sccenc/sccparse: Scenarist Closed Caption format encoder and parser
-- dav1dec: AV1 video decoder based on the dav1d decoder implementation
- by the VLC project
-- rav1enc: AV1 video encoder based on the fast and pure-Rust rav1e
- encoder implementation
-- rsflvdemux: Alternative to the flvdemux FLV demuxer element from
- gst-plugins-good, not feature-equivalent yet
-- rsgifenc/rspngenc: GIF/PNG encoder elements based on the pure-Rust
- implementations by the image-rs project
-
-Rust text plugins
-
-- textwrap: Element for line-wrapping timed text (e.g. subtitles) for
- better screen-fitting, including hyphenation support for some
- languages
-
-Rust network plugins
-
-- reqwesthttpsrc: HTTP(S) source element based on the Rust
- reqwest/hyper HTTP implementations and almost feature-equivalent
- with the main GStreamer HTTP source souphttpsrc
-- s3src/s3sink: Source/sink element for the Amazon S3 cloud storage
-- awstranscriber: Live audio to timed text transcription element using
- the Amazon AWS Transcribe API
-
-Generic Rust plugins
-
-- sodiumencrypter/sodiumdecrypter: Encryption/decryption element based
- on libsodium/NaCl
-- togglerecord: Recording element that allows to pause/resume
- recordings easily and considers keyframe boundaries
-- fallbackswitch/fallbacksrc: Elements for handling potentially
- failing (network) sources, restarting them on errors/timeout and
- showing a fallback stream instead
-- threadshare: Set of elements that provide alternatives for various
- existing GStreamer elements but allow to share the streaming threads
- between each other to reduce the number of threads
-- rsfilesrc/rsfilesink: File source/sink elements as replacements for
- the existing filesrc/filesink elements
+- The GStreamer Rust bindings are released separately with a different
+ release cadence that’s tied to gtk-rs, but the latest release has
+ already been updated for the upcoming new GStreamer 1.20 API (v1_20
+ feature).
+
+- gst-plugins-rs, the module containing GStreamer plugins written in
+ Rust, has also seen lots of activity with many new elements and
+ plugins. See the New Elements section above for a list of new Rust
+ elements.
Build and Dependencies
- Meson 0.59 or newer is required to build GStreamer now.
-- FIXME: this section will be filled in in due course
+- The GLib requirement has been bumped to GLib 2.56 or newer (from
+ March 2018).
+
+- The wpe plugin now requires wpe >= 2.28 and wpebackend-fdo >= 1.8
Explicit opt-in required for build of certain plugins with (A)GPL dependencies
See Building plugins with (A)GPL-licensed dependencies for more details
and a non-exhaustive list of plugins affected.
-gst-build: replaced by Monorepo
+gst-build: replaced by mono repository
-- this section will be filled in in due course
-
-- FIXME: describe + link to Monorepo FAQ
+See mono repository section above and the GStreamer mono repository FAQ.
Cerbero
on platforms where dependencies are not readily available, such as
Windows, Android, iOS and macOS.
-General improvements
+General Cerbero improvements
-- this section will be filled in in due course
+- Plugin removed: libvisual
+- New plugins: rtpmanagerbad and rist
-macOS / iOS
+macOS / iOS specific Cerbero improvements
-- this section will be filled in in due course
+- XCode 12 support
+- macOS OS release support is now future-proof, similar to iOS
+- macOS Apple Silicon (ARM64) cross-compile support has been added
+- macOS Apple Silicon (ARM64) native support is currently experimental
-Windows
+Windows specific Cerbero improvements
-- this section will be filled in in due course
+- Visual Studio 2022 support has been added
+- bootstrap is faster since it requires building fewer build-tools
+ recipes on Windows
+- package is faster due to better scheduling of recipe stages and
+ elimination of unnecessary autotools regeneration
+- The following plugins are no longer built on Windows:
+ - a52dec (another decoder is still available in libav)
+ - dvdread
+ - resindvd
Windows MSI installer
-- this section will be filled in in due course
+- no major changes
-Linux
+Linux specific Cerbero improvements
-- this section will be filled in in due course
+- Fedora, Debian OS release support is now more future-proof
+- Amazon Linux 2 support has been added
-Android
+Android specific Cerbero improvements
-- this section will be filled in in due course
+- no major changes
Platform-specific changes and improvements
Android
-- this section will be filled in in due course
+- No major changes
macOS and iOS
-- this section will be filled in in due course
+- applemedia: add ProRes support to vtenc and vtdec
Windows
-- this section will be filled in in due course
+- On Windows the high-resolution clock is enabled now in the
+ gst-launch-1.0 and gst-play-1.0 command line tools, which provides
+ better clock and timer performance on Windows, at the cost of higher
+ power consumption. By default, without the high-resolution clock
+ enabled, the timer precision on Windows is system-dependent and may
+ be as bad as 15ms which is not good enough for many multimedia
+ applications. Developers may want to do the same in their Windows
+ applications if they think it’s a good idea for their application
+ use case, and depending on the Windows version they target. This is
+ not done automatically by GStreamer because on older Windows
+ versions (pre-Windows 10) this affects a global Windows setting and
+ also there’s a power consumption vs. performance trade-off that may
+ differ from application to application.
+
+- dxgiscreencapsrc now supports resolution changes
+
+- The wasapi2 audio plugin was rewritten and now has a higher rank
+ than the old wasapi plugin since it has a number of additional
+ features such as automatic stream routing, and no
+ known-but-hard-to-fix issues. The plugin is always built if the
+ Windows 10 SDK is available now.
+
+- The wasapi device providers now detect and notify dynamic device
+ additions/removals
+
+- d3d11screencapturesrc: new desktop capture element, including
+ GstDeviceProvider implementation to enumerate/select target monitors
+ for capture.
+
+- Direct3D11/DXVA decoder now supports AV1 and MPEG2 codecs
+ (d3d11av1dec, d3d11mpeg2dec)
+
+- VP9 decoding got more reliable and stable thanks to a newly written
+ codec parser
+
+- Support for decoding interlaced H.264/AVC streams
+
+- Hardware-accelerated video deinterlacing (d3d11deinterlace) and
+ video mixing (d3d11compositor)
+
+- Video mixing with the Direct3D11 API (d3d11compositor)
+
+- MediaFoundation API based hardware encoders gained the ability to
+ receive Direct3D11 textures as an input
+
+- Seungha’s blog post “GStreamer ❤ Windows: A primer on the cool stuff
+ you’ll find in the 1.20 release” describes many of the
+ Windows-related improvements in more detail
Linux
-- this section will be filled in in due course
+- bluez: LDAC Bluetooth audio codec support in a2dpsink and avdtpsink,
+ as well as an LDAC RTP payloader (rtpldacpay) and an LDAC audio
+ encoder (ldacenc)
+
+- kmssink: gained support for NV24, NV61, RGB16/BGR16 formats;
+ auto-detect NVIDIA Tegra driver
Documentation improvements
-- this section will be filled in in due course
+- hardware-accelerated GPU plugins will now no longer always list all
+ the element variants for all available GPUs, since those are
+ system-dependent and it’s confusing for users to see those in the
+ documentation just because the GStreamer developer who generated the
+ docs had multiple GPUs to play with at the time. Instead just show
+ the default elements.
-Possibly Breaking Changes
+Possibly Breaking and Other Noteworthy Behavioural Changes
+
+- gst_parse_launch(), gst_parse_bin_from_description() and friends
+ will now error out when setting properties that don’t exist on
+ top-level bins. They were silently ignored before.
+
+- The GstWebRTC library does not expose any objects anymore with
+ public fields. Instead properties have been added to replace that
+ functionality. If you are accessing such fields in your application,
+ switch to the corresponding properties.
-- FIXME: this section will be filled in in due course
-- MPEG-TS SCTE-35 API changes (FIXME: flesh out)
-- gst_parse_launch() and friends now error out on non-existing
- properties on top-level bins where they would silently fail and
- ignore those before.
- playbin and uridecodebin now emit the source-setup signal before the
element is added to the bin and linked so that the source element is
already configured before any scheduling query comes in, which is
- useful for elements such as appsrc or giostreamsrc. (Merge Request)
+ useful for elements such as appsrc or giostreamsrc.
+
+- The source element inside urisourcebin (used inside uridecodebin3
+ which is used inside playbin3) is no longer called "source". This
+ shouldn’t affect anyone hopefully, because there’s a "setup-source"
+ signal to configure the source element and no one should rely on
+ names of internal elements anyway.
+
+- The vp8enc element now expects bps (bits per second) for the
+ "temporal-scalability-target-bitrate" property, which is consistent
+ with the "target-bitrate" property. Since additional configuration
+ is required with modern libvpx to make temporal scaling work anyway,
+ chances are that very few people will have been using this property
+
+- vp8enc and vp9enc now default to “good quality” for the "deadline"
+ property rather then “best quality”. Having the deadline set to best
+ quality causes the encoder to be absurdly slow, most real-life users
+ will want the good quality tradeoff instead.
+
+- The experimental GstTranscoder library API in gst-plugins-bad was
+ changed from a GObject signal-based notification mechanism to a
+ GstBus/message-based mechanism akin to GstPlayer/GstPlay.
+
+- MPEG-TS SCTE-35 API: semantic change for SCTE-35 splice commands:
+ timestamps passed by the application should be in running time now,
+ since users of the API can’t really be expected to predict the local
+ PTS of the muxer.
+
+- The GstContext used by souphttpsrc to share the session between
+ multiple element instances has changed. Previously it provided
+ direct access to the internal SoupSession object, now it only
+ provides access to an opaque, internal type. This change is
+ necessary because SoupSession is not thread-safe at all and can’t be
+ shared safely between arbitrary external code and souphttpsrc.
+
+- Python bindings: GObject-introspection related Annotation fixes have
+ led to a case of a GstVideo.VideoInfo-related function signature
+ changing in the Python bindings (possibly one or two other cases
+ too). This is for a function that should never have been exposed in
+ the first place though, so the bindings are being updated to throw
+ an exception in that case, and the correct replacement API has been
+ added in form of an override.
Known Issues
-- this section will be filled in in due course
-
-- There are a couple of known WebRTC-related regressions/blockers:
-
- - webrtc: DTLS setup with Chrome is broken
- - webrtcbin: First keyframe is usually lost
+- nothing in particular at this point (but also see possibly breaking
+ changes section above)
Contributors
-- this section will be filled in in due course
+Aaron Boxer, Adam Leppky, Adam Williamson, Alba Mendez, Alejandro
+González, Aleksandr Slobodeniuk, Alexander Vandenbulcke, Alex Ashley,
+Alicia Boya García, Andika Triwidada, Andoni Morales Alastruey, Andrew
+Wesie, Andrey Moiseev, Antonio Ospite, Antonio Rojas, Arthur Crippa
+Búrigo, Arun Raghavan, Ashley Brighthope, Axel Kellermann, Baek, Bastien
+Nocera, Bastien Reboulet, Benjamin Gaignard, Bing Song, Binh Truong,
+Biswapriyo Nath, Brad Hards, Brad Smith, Brady J. Garvin, Branko
+Subasic, Camilo Celis Guzman, Chris Bass, ChrisDuncanAnyvision, Chris
+White, Corentin Damman, Daniel Almeida, Daniel Knobe, Daniel Stone,
+david, David Fernandez, David Keijser, David Phung, Devarsh Thakkar,
+Dinesh Manajipet, Dmitry Samoylov, Dmitry Shusharin, Dominique Martinet,
+Doug Nazar, Ederson de Souza, Edward Hervey, Emmanuel Gil Peyrot,
+Enrique Ocaña González, Ezequiel Garcia, Fabian Orccon, Fabrice
+Fontaine, Fernando Jimenez Moreno, Florian Karydes, Francisco Javier
+Velázquez-García, François Laignel, Frederich Munch, Fredrik Pålsson,
+George Kiagiadakis, Georg Lippitsch, Göran Jönsson, Guido Günther,
+Guillaume Desmottes, Guiqin Zou, Haakon Sporsheim, Haelwenn (lanodan)
+Monnier, Haihao Xiang, Haihua Hu, Havard Graff, He Junyan, Helmut
+Januschka, Henry Wilkes, Hosang Lee, Hou Qi, Ignacio Casal Quinteiro,
+Igor Kovalenko, Ilya Kreymer, Imanol Fernandez, Jacek Tomaszewski, Jade
+Macho, Jakub Adam, Jakub Janků, Jan Alexander Steffens (heftig), Jan
+Schmidt, Jason Carrete, Jason Pereira, Jay Douglass, Jeongki Kim, Jérôme
+Laheurte, Jimmi Holst Christensen, Johan Sternerup, John Hassell, John
+Lindgren, John-Mark Bell, Jonathan Matthew, Jordan Petridis, Jose
+Quaresma, Julian Bouzas, Julien, Kai Uwe Broulik, Kasper Steensig
+Jensen, Kellermann Axel, Kevin Song, Khem Raj, Knut Inge Hvidsten, Knut
+Saastad, Kristofer Björkström, Lars Lundqvist, Lawrence Troup, Lim Siew
+Hoon, Lucas Stach, Ludvig Rappe, Luis Paulo Fernandes de Barros, Luke
+Yelavich, Mads Buvik Sandvei, Marc Leeman, Marco Felsch, Marek Vasut,
+Marian Cichy, Marijn Suijten, Marius Vlad, Markus Ebner, Mart Raudsepp,
+Matej Knopp, Mathieu Duponchelle, Matthew Waters, Matthieu De Beule,
+Mengkejiergeli Ba, Michael de Gans, Michael Olbrich, Michael Tretter,
+Michal Dzik, Miguel Paris, Mikhail Fludkov, mkba, Nazar Mokrynskyi,
+Nicholas Jackson, Nicola Murino, Nicolas Dufresne, Niklas Hambüchen,
+Nikolay Sivov, Nirbheek Chauhan, Olivier Blin, Olivier Crete, Olivier
+Crête, Paul Goulpié, Per Förlin, Peter Boba, P H, Philippe Normand,
+Philipp Zabel, Pieter Willem Jordaan, Piotrek Brzeziński, Rafał
+Dzięgiel, Rafostar, raghavendra, Raghavendra, Raju Babannavar, Raleigh
+Littles III, Randy Li, Randy Li (ayaka), Ratchanan Srirattanamet, Raul
+Tambre, reed.lawrence, Ricky Tang, Robert Rosengren, Robert Swain, Robin
+Burchell, Roman Sivriver, R S Nikhil Krishna, Ruben Gonzalez, Ruslan
+Khamidullin, Sanchayan Maity, Scott Moreau, Sebastian Dröge, Sergei
+Kovalev, Seungha Yang, Sid Sethupathi, sohwan.park, Sonny Piers, Staz M,
+Stefan Brüns, Stéphane Cerveau, Stephan Hesse, Stian Selnes, Stirling
+Westrup, Théo MAILLART, Thibault Saunier, Tim, Timo Wischer, Tim-Philipp
+Müller, Tim Schneider, Tobias Ronge, Tom Schoonjans, Tulio Beloqui,
+tyler-aicradle, U. Artie Eoff, Ung, Val Doroshchuk, VaL Doroshchuk,
+Víctor Manuel Jáquez Leal, Vivek R, Vivia Nikolaidou, Vivienne
+Watermeier, Vladimir Menshakov, Will Miller, Wim Taymans, Xabier
+Rodriguez Calvar, Xavier Claessens, Xℹ Ruoyao, Yacine Bandou, Yinhang
+Liu, youngh.lee, youngsoo.lee, yychao, Zebediah Figura, Zhang yuankun,
+Zhang Yuankun, Zhao, Zhao Zhili, , Aleksandar Topic, Antonio Ospite,
+Bastien Nocera, Benjamin Gaignard, Brad Hards, Carlos Falgueras García,
+Célestin Marot, Corentin Damman, Corentin Noël, Daniel Almeida, Daniel
+Knobe, Danny Smith, Dave Piché, Dmitry Osipenko, Fabrice Fontaine,
+fjmax, Florian Zwoch, Guillaume Desmottes, Haihua Hu, Heinrich Kruger,
+He Junyan, Jakub Adam, James Cowgill, Jan Alexander Steffens (heftig),
+Jean Felder, Jeongki Kim, Jiri Uncovsky, Joe Todd, Jordan Petridis,
+Krystian Wojtas, Marc-André Lureau, Marcin Kolny, Marc Leeman, Mark
+Nauwelaerts, Martin Reboredo, Mathieu Duponchelle, Matthew Waters,
+Mengkejiergeli Ba, Michael Gruner, Nicolas Dufresne, Nirbheek Chauhan,
+Olivier Crête, Philippe Normand, Rafał Dzięgiel, Ralf Sippl, Robert
+Mader, Sanchayan Maity, Sangchul Lee, Sebastian Dröge, Seungha Yang,
+Stéphane Cerveau, Teh Yule Kim, Thibault Saunier, Thomas Klausner, Timo
+Wischer, Tim-Philipp Müller, Tobias Reineke, Tomasz Andrzejak, Trung Do,
+Tyler Compton, Ung, Víctor Manuel Jáquez Leal, Vivia Nikolaidou, Wim
+Taymans, wngecn, Wonchul Lee, wuchang li, Xavier Claessens, Xi Ruoyao,
+Yoshiharu Hirose, Zhao,
… and many others who have contributed bug reports, translations, sent
suggestions or helped testing.
1.20.0
-1.20.0 is scheduled to be released around October/November 2021.
+1.20.0 is scheduled to be released around early February 2022.
Schedule for 1.22
unstable development version leading up to the stable 1.22 release. The
development of 1.21/1.22 will happen in the git main branch.
-The plan for the 1.22 development cycle is yet to be confirmed.
+The plan for the 1.22 development cycle is yet to be confirmed. Assuming
+no major project-wide reorganisations in the 1.22 cycle we might try and
+aim for a release around August 2022.
1.22 will be backwards-compatible to the stable 1.20, 1.18, 1.16, 1.14,
1.12, 1.10, 1.8, 1.6, 1.4, 1.2 and 1.0 release series.
------------------------------------------------------------------------
These release notes have been prepared by Tim-Philipp Müller with
-contributions from …
+contributions from Matthew Waters, Nicolas Dufresne, Nirbheek Chauhan,
+Sebastian Dröge and Seungha Yang.
License: CC BY-SA 4.0
-This is GStreamer gstreamer-vaapi 1.19.3.
+This is GStreamer gstreamer-vaapi 1.19.90.
GStreamer 1.19 is the development branch leading up to the next major
stable version which will be 1.20.
<release>
<Version>
+ <revision>1.19.90</revision>
+ <branch>main</branch>
+ <name></name>
+ <created>2022-01-28</created>
+ <file-release rdf:resource="https://gstreamer.freedesktop.org/src/gstreamer-vaapi/gstreamer-vaapi-1.19.90.tar.xz" />
+ </Version>
+ </release>
+
+ <release>
+ <Version>
<revision>1.19.3</revision>
<branch>main</branch>
<name></name>
project('gstreamer-vaapi', 'c',
- version : '1.19.3.1',
+ version : '1.19.90',
meson_version : '>= 0.59',
default_options : [ 'warning_level=1',
'buildtype=debugoptimized' ])
+=== release 1.19.90 ===
+
+2022-01-28 14:28:35 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * NEWS:
+ * RELEASE:
+ * gstreamer.doap:
+ * meson.build:
+ Release 1.19.90
+
+2022-01-28 14:28:28 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * ChangeLog:
+ Update ChangeLogs for 1.19.90
+
2021-12-23 00:24:03 +0100 Mathieu Duponchelle <mathieu@centricular.com>
* libs/gst/base/gstaggregator.c:
GStreamer 1.20 Release Notes
GStreamer 1.20 has not been released yet. It is scheduled for release in
-November 2021.
+late January / early February 2022.
1.19.x is the unstable development version that is being developed in
-the git main branch and which will eventually result in 1.20, and 1.19.3
-is the current development release in that series
-
-It is expected that feature freeze will be in early October 2021,
-followed by one or two 1.19.9x pre-releases and the new 1.20 stable
-release around October/November 2021.
+the git main branch and which will eventually result in 1.20, and
+1.19.90 is the first release candidate in that series (1.20rc1).
1.20 will be backwards-compatible to the stable 1.18, 1.16, 1.14, 1.12,
-1.10, 1.8, 1.6,, 1.4, 1.2 and 1.0 release series.
+1.10, 1.8, 1.6, 1.4, 1.2 and 1.0 release series.
See https://gstreamer.freedesktop.org/releases/1.20/ for the latest
version of this document.
-Last updated: Monday 1 November 2021, 01:00 UTC (log)
+Last updated: Wednesday 26 January 2022, 01:00 UTC (log)
Introduction
Highlights
-- this section will be completed in due course
+- Development in GitLab was switched to a single git repository
+ containing all the modules
+- GstPlay: new high-level playback library, replaces GstPlayer
+- WebM Alpha decoding support
+- Encoding profiles can now be tweaked with additional
+ application-specified element properties
+- Compositor: multi-threaded video conversion and mixing
+- RTP header extensions: unified support in RTP depayloader and
+ payloader base classes
+- SMPTE 2022-1 2-D Forward Error Correction support
+- Smart encoding (passthrough) support for VP8, VP9, H.265 in
+ encodebin and transcodebin
+- Runtime compatibility support for libsoup2 and libsoup3 (libsoup3
+ support experimental)
+- Video decoder subframe support
+- Video decoder automatic packet-loss, data corruption, and keyframe
+ request handling for RTP / WebRTC / RTSP
+- MP4 and Matroska muxers now support profile/level/resolution changes
+ for H264/H265 input streams (i.e. codec data changing on the fly)
+- MP4 muxing mode that initially creates a fragmented mp4 which is
+ converted to a regular mp4 on EOS
+- Audio support for the WebKit Port for Embedded (WPE) web page source
+ element
+- CUDA based video color space convert and rescale elements and
+ upload/download elements
+- NVIDIA memory:NVMM support for OpenGL glupload and gldownload
+ elements
+- Many WebRTC improvements
+- The new VA-API plugin implemention fleshed out with more decoders
+ and new postproc elements
+- AppSink API to retrieve events in addition to buffers and buffer
+ lists
+- AppSrc gained more configuration options for the internal queue
+ (leakiness, limits in buffers and time, getters to read current
+ levels)
+- Updated Rust bindings and many new Rust plugins
+- Improved support for custom minimal GStreamer builds
+- Support build against FFmpeg 5.0
+- Linux Stateless CODEC support gained MPEG2 and VP9
+- Windows Direct3D11/DXVA decoder gained AV1 and MPEG2 support
+- Lots of new plugins, features, performance improvements and bug
+ fixes
Major new features and changes
Noteworthy new features and API
-- this section will be filled in in due course
+- gst_element_get_request_pad() has been deprecated in favour of the
+ newly-added gst_element_request_pad_simple() which does the exact
+ same thing but has a less confusing name that hopefully makes clear
+ that the function request a new pad rather than just retrieves an
+ already-existing request pad.
+
+Development in GitLab was switched to a single git repository containing all the modules
+
+The GStreamer multimedia framework is a set of libraries and plugins
+split into a number of distinct modules which are released independently
+and which have so far been developed in separate git repositories in
+freedesktop.org GitLab.
+
+In addition to these separate git repositories there was a gst-build
+module that would use the Meson build systems’s subproject feature to
+download each individual module and then build everything in one go. It
+would also provide an uninstalled development environment that made it
+easy to work on GStreamer and use or test versions other than the
+system-installed GStreamer version.
+
+All of these modules have now (as of 28 September 2021) been merged into
+a single git repository (“Mono repository” or “monorepo”) which should
+simplify development workflows and continuous integration, especially
+where changes need to be made to multiple modules at once.
+
+This mono repository merge will primarily affect GStreamer developers
+and contributors and anyone who has workflows based on the GStreamer git
+repositories.
+
+The Rust bindings and Rust plugins modules have not been merged into the
+mono repository at this time because they follow a different release
+cycle.
+
+The mono repository lives in the existing GStreamer core git repository
+in GitLab in the new main branch and all future development will happen
+on this branch.
+
+Modules will continue to be released as separate tarballs.
+
+For more details, please see the GStreamer mono repository FAQ.
+
+GstPlay: new high-level playback library replacing GstPlayer
+
+- GstPlay is a new high-level playback library that replaces the older
+ GstPlayer API. It is basically the same API as GstPlayer but
+ refactored to use bus messages for application notifications instead
+ of GObject signals. There is still a signal adapter object for those
+ who prefer signals. Since the existing GstPlayer API is already in
+ use in various applications, it didn’t seem like a good idea to
+ break it entirely. Instead a new API was added, and it is expected
+ that this new GstPlay API will be moved to gst-plugins-base in
+ future.
+
+- The existing GstPlayer API is scheduled for deprecation and will be
+ removed at some point in the future (e.g. in GStreamer 1.24), so
+ application developers are urged to migrate to the new GstPlay API
+ at their earliest convenience.
+
+WebM alpha decoding
+
+- Implement WebM alpha decoding (VP8/VP9 with alpha), which required
+ support and additions in various places. This is supported both with
+ software decoders and hardware-accelerated decoders.
+
+- VP8/VP9 don’t support alpha components natively in the codec, so the
+ way this is implemented in WebM is by encoding the alpha plane with
+ transparency data as a separate VP8/VP9 stream. Inside the WebM
+ container (a variant of Matroska) this is coded as a single video
+ track with the “normal” VP8/VP9 video data making up the main video
+ data and each frame of video having an encoded alpha frame attached
+ to it as extra data ("BlockAdditional").
+
+- matroskademux has been extended extract this per-frame alpha side
+ data and attach it in form of a GstVideoCodecAlphaMeta to the
+ regular video buffers. Note that this new meta is specific to this
+ VP8/VP9 alpha support and can’t be used to just add alpha support to
+ other codecs that don’t support it. Lastly, matroskademux also
+ advertises the fact that the streams contain alpha in the caps.
+
+- The new codecalpha plugin contains various bits of infrastructure to
+ support autoplugging and debugging:
+
+ - codecalphademux splits out the alpha stream from the metas on
+ the regular VP8/VP9 buffers
+ - alphacombine takes two decoded raw video streams (one alpha, one
+ the regular video) and combines it into a video stream with
+ alpha
+ - vp8alphadecodebin + vp9alphadecodebin are wrapper bins that use
+ the regular vp8dec and vp9dec software decoders to decode
+ regular and alpha streams and combine them again. To decodebin
+ these look like regular decoders which ju
+ - The V4L2 CODEC plugin has stateless VP8/VP9 decoders that can
+ decode both alpha and non-alpha stream with a single decoder
+ instance
+
+- A new AV12 video format was added which is basically NV12 with an
+ alpha plane, which is more convenient for many hardware-accelerated
+ decoders.
+
+- Watch Nicolas Dufresne’s LCA 2022 talk “Bringing WebM Alpha support
+ to GStreamer” for all the details and a demo.
+
+RTP Header Extensions Base Class and Automatic Header Extension Handling in RTP Payloaders and Depayloaders
+
+- RTP Header Extensions are specified in RFC 5285 and provide a way to
+ add small pieces of data to RTP packets in between the RTP header
+ and the RTP payload. This is often used for per-frame metadata,
+ extended timestamps or other application-specific extra data. There
+ are several commonly-used extensions specified in various RFCs, but
+ senders are free to put any kind of data in there, as long as sender
+ and receiver both know what that data is. Receivers that don’t know
+ about the header extensions will just skip the extra data without
+ ever looking at it. These header extensions can often be combined
+ with any kind of payload format, so may need to be supported by many
+ RTP payloader and depayloader elements.
+
+- Inserting and extracting RTP header extension data has so far been a
+ bit inconvenient in GStreamer: There are functions to add and
+ retrieve RTP header extension data from RTP packets, but nothing
+ works automatically, even for common extensions. People would have
+ to do the insertion/extraction either in custom elements
+ before/after the RTP payloader/depayloader, or inside pad probes,
+ which isn’t very nice.
+
+- This release adds various pieces of new infrastructure for generic
+ RTP header extension handling, as well as some implementations for
+ common extensions:
+
+ - GstRTPHeaderExtension is a new helper base class for reading and
+ writing RTP header extensions. Nominally this subclasses
+ GstElement, but only so these extensions are stored in the
+ registry where they can be looked up by URI or name. They don’t
+ have pads and don’t get added to the pipeline graph as an
+ element.
+
+ - "add-extension" and "clear-extension" action signals on RTP
+ payloaders and depayloaders for manual extension management
+
+ - The "request-extension" signal will be emitted if an extension
+ is encountered that requires explicit mapping by the application
+
+ - new "auto-header-extension" property on RTP payloaders and
+ depayloaders for automatic handling of known header extensions.
+ This is enabled by default. The extensions must be signalled via
+ caps / SDP.
+
+ - RTP header extension implementations:
+
+ - rtphdrextclientaudiolevel: Client-to-Mixer Audio Level
+ Indication (RFC 6464) (also see below)
+ - rtphdrextcolorspace: Color Space extension, extends RTP
+ packets with color space and high dynamic range (HDR)
+ information
+ - rtphdrexttwcc: Transport Wide Congestion Control support
+
+- gst_rtp_buffer_remove_extension_data() is a new helper function to
+ remove an RTP header extension from an RTP buffer
+
+- The existing gst_rtp_buffer_set_extension_data() now also supports
+ shrinking the extension data in size
+
+AppSink and AppSrc improvements
+
+- appsink: new API to pull events out of appsink in addition to
+ buffers and buffer lists.
+
+ There was previously no way for users to receive incoming events
+ from appsink properly serialised with the data flow, even if they
+ are serialised events. The reason for that is that the only way to
+ intercept events was via a pad probe on the appsink sink pad, but
+ there is also internal queuing inside of appsink, so it’s difficult
+ to ascertain the right order of everything in all cases.
+
+ There is now a new "new-serialized-event" signal which will be
+ emitted when there’s a new event pending (just like the existing
+ "new-sample" signal). The "emit-signals" property must be set to
+ TRUE in order to activate this (but it’s also fine to just pull from
+ the application thread without using the signals).
+
+ gst_app_sink_pull_object() and gst_app_sink_try_pull_object() can be
+ used to pull out either an event or a new sample carrying a buffer
+ or buffer list, whatever is next in the queue.
+
+ EOS events will be filtered and will not be returned. EOS handling
+ can be done the ususal way, same as with _pull_sample().
+
+- appsrc: allow configuration of internal queue limits in time and
+ buffers and add leaky mode.
+
+ There is internal queuing inside appsrc so the application thread
+ can push data into the element which will then be picked up by the
+ source element’s streaming thread and pushed into the pipeline from
+ that streaming thread. This queue is unlimited by default and until
+ now it was only possible to set a maximum size limit in bytes. When
+ that byte limit is reached, the pushing thread (application thread)
+ would be blocked until more space becomes available.
+
+ A limit in bytes is not particularly useful for many use cases, so
+ now it is possible to also configure limits in time and buffers
+ using the new "max-time" and "max-buffers" properties. Of course
+ there are also matching new read-only"current-level-buffers" and
+ "current-level-time properties" properties to query the current fill
+ level of the internal queue in time and buffers.
+
+ And as if that wasn’t enough the internal queue can also be
+ configured as leaky using the new "leaky-type" property. That way
+ when the queue is full the application thread won’t be blocked when
+ it tries to push in more data, but instead either the new buffer
+ will be dropped or the oldest data in the queue will be dropped.
+
+Better string serialization of nested GstCaps and GstStructures
+
+- New string serialisation format for structs and caps that can handle
+ nested structs and caps properly by using brackets to delimit nested
+ items (e.g. some-struct, some-field=[nested-struct, nested=true]).
+ Unlike the default format the new variant can also support more than
+ one level of nesting. For backwards-compatibility reasons the old
+ format is still output by default when serialising caps and structs
+ using the existing API. The new functions gst_caps_serialize() and
+ gst_structure_serialize() can be used to output strings in the new
+ format.
+
+Convenience API for custom GstMetas
+
+- New convenience API to register and create custom GstMetas:
+ gst_meta_register_custom() and gst_buffer_add_custom_meta(). Such
+ custom meta is backed by a GstStructure and does not require that
+ users of the API expose their GstMeta implementation as public API
+ for other components to make use of it. In addition, it provides a
+ simpler interface by ignoring the impl vs. api distinction that the
+ regular API exposes. This new API is meant to be the meta
+ counterpart to custom events and messages, and to be more convenient
+ than the lower-level API when the absolute best performance isn’t a
+ requirement. The reason it’s less performant than a “proper” meta is
+ that a proper meta is just a C struct in the end whereas this goes
+ through the GstStructure API which has a bit more overhead, which
+ for most scenarios is negligible however. This new API is useful for
+ experimentation or proprietary metas, but also has some limitations:
+ it can only be used if there’s a single producer of these metas;
+ it’s not allowed to register the same custom meta multiple times or
+ from multiple places.
+
+Additional Element Properties on Encoding Profiles
+
+- GstEncodingProfile: The new "element-properties" and
+ gst_encoding_profile_set_element_properties() API allows
+ applications to set additional element properties on encoding
+ profiles to configure muxers and encoders. So far the encoding
+ profile template was the only place where this could be specified,
+ but often what applications want to do is take a ready-made encoding
+ profile shipped by GStreamer or the application and then tweak the
+ settings on top of that, which is now possible with this API. Since
+ applications can’t always know in advance what encoder element will
+ be used in the end, it’s even possible to specify properties on a
+ per-element basis.
+
+ Encoding Profiles are used in the encodebin, transcodebin and
+ camerabin elements and APIs to configure output formats (containers
+ and elementary streams).
+
+Audio Level Indication Meta for RFC 6464
+
+- New GstAudioLevelMeta containing Audio Level Indication as per RFC
+ 6464
+
+- The level element has been updated to add GstAudioLevelMeta on
+ buffers if the "audio-level-meta" property is set to TRUE. This can
+ then in turn be picked up by RTP payloaders to signal the audio
+ level to receivers through RTP header extensions (see above).
+
+- New Client-to-Mixer Audio Level Indication (RFC6464) RTP Header
+ Extension which should be automatically created and used by RTP
+ payloaders and depayloaders if their "auto-header-extension"
+ property is enabled and if the extension is part of the RTP caps.
+
+Automatic packet loss, data corruption and keyframe request handling for video decoders
+
+- The GstVideoDecoder base class has gained various new APIs to
+ automatically handle packet loss and data corruption better by
+ default, especially in RTP, RTSP and WebRTC streaming scenarios, and
+ to give subclasses more control about how they want to handle
+ missing data:
+
+ - Video decoder subclasses can mark output frames as corrupted via
+ the new GST_VIDEO_CODEC_FRAME_FLAG_CORRUPTED flag
+
+ - A new "discard-corrupted-frames" property allows applications to
+ configure decoders so that corrupted frames are directly
+ discarded instead of being forwarded inside the pipeline. This
+ is a replacement for the "output-corrupt" property of the FFmpeg
+ decoders.
+
+ - RTP depayloaders can now signal to decoders that data is missing
+ when sending GAP events for lost packets. GAP events can be sent
+ for various reason in a GStreamer pipeline. Often they are just
+ used to let downstream elements know that there isn’t a buffer
+ available at the moment, so downstream elements can move on
+ instead of waiting for one. They are also sent by RTP
+ depayloaders in the case that packets are missing, however, and
+ so far a decoder was not able to differentiate the two cases.
+ This has been remedied now: GAP events can be decorated with
+ gst_event_set_gap_flags() and GST_GAP_FLAG_MISSING_DATA to let
+ decoders now what happened, and decoders can then use that in
+ some cases to handle missing data better.
+
+ - The GstVideoDecoder::handle_missing_data vfunc was added to
+ inform subclasses about packet loss or missing data and let them
+ handle it in their own way if they like.
+
+ - gst_video_decoder_set_needs_sync_point() lets subclasses signal
+ that they need the stream to start with a sync point. If
+ enabled, the base class will discard all non-sync point frames
+ in the beginning and after a flush and does not pass them to the
+ subclass. Furthermore, if the first frame is not a sync point,
+ the base class will try and request a sync frame from upstream
+ by sending a force-key-unit event (see next items).
+
+ - New "automatic-request-sync-points" and
+ "automatic-request-sync-point-flags" properties to automatically
+ request sync points when needed, e.g. on packet loss or if the
+ first frame is not a keyframe. Applications may want to enable
+ this on decoders operating in e.g. RTP/WebRTC/RTSP receiver
+ pipelines.
+
+ - The new "min-force-key-unit-interval" property can be used to
+ ensure there’s a minimal interval between keyframe requests to
+ upstream (and/or the sender) and we’re not flooding the sender
+ with key unit requests.
+
+ - gst_video_decoder_request_sync_point() allows subclasses to
+ request a new sync point (e.g. if they choose to do their own
+ missing data handling). This will still honour the
+ "min-force-key-unit-interval" property if set.
+
+Improved support for custom minimal GStreamer builds
+
+- Element registration and registration of other plugin features
+ inside plugin init functions has been improved in order to
+ facilitate minimal custom GStreamer builds.
+
+- A number of new macros have been added to declare and create
+ per-element and per-pluginfeature register functions in all plugins,
+ and then call those from the per-plugin plugin_init functions:
+
+ - GST_ELEMENT_REGISTER_DEFINE,
+ GST_DEVICE_PROVIDER_REGISTER_DEFINE,
+ GST_DYNAMIC_TYPE_REGISTER_DEFINE, GST_TYPE_FIND_REGISTER_DEFINE
+ for the actual registration call with GStreamer
+ - GST_ELEMENT_REGISTER, GST_DEVICE_PROVIDER_REGISTER,
+ GST_DYNAMIC_TYPE_REGISTER, GST_PLUGIN_STATIC_REGISTER,
+ GST_TYPE_FIND_REGISTER to call the registration function defined
+ by the REGISTER_DEFINE macro
+ - GST_ELEMENT_REGISTER_DECLARE,
+ GST_DEVICE_PROVIDER_REGISTER_DECLARE,
+ GST_DYNAMIC_TYPE_REGISTER_DECLARE,
+ GST_TYPE_FIND_REGISTER_DECLARE to declare the registration
+ function defined by the REGISTER_DEFINE macro
+ - and various variants for advanced use cases.
+
+- This means that applications can call the per-element and
+ per-pluginfeature registration functions for only the elements they
+ need instead of registering plugins as a whole with all kinds of
+ elements that may not be required (e.g. encoder and decoder instead
+ of just decoder). In case of static linking all unused functions and
+ their dependencies would be removed in this case by the linker,
+ which helps minimise binary size for custom builds.
+
+- gst_init() will automatically call a gst_init_static_plugins()
+ function if one exists.
+
+- See the GStreamer static build documentation and Stéphane’s blog
+ post Generate a minimal GStreamer build, tailored to your needs for
+ more details.
New elements
-- this section will be filled in in due course
+- New aesdec and aesenc elements for AES encryption and decryption in
+ a custom format.
+
+- New encodebin2 element with dynamic/sometimes source pads in order
+ to support the option of doing the muxing outside of encodebin,
+ e.g. in combination with a splitmuxsink.
+
+- New fakeaudiosink and videocodectestsink elements for testing and
+ debugging (see below for more details)
+
+- rtpisacpay, rtpisacdepay: new RTP payloader and depayloader for iSAC
+ audio codec
+
+- rtpst2022-1-fecdec, rtpst2022-1-fecenc: new elements providing SMPTE
+ 2022-1 2-D Forward Error Correction. More details in Mathieu’s blog
+ post.
+
+- isac: new plugin wrapping the Internet Speech Audio Codec reference
+ encoder and decoder from the WebRTC project.
+
+- asio: plugin for Steinberg ASIO (Audio Streaming Input/Output) API
+
+- gssrc, gssink: add source and sink for Google Cloud Storage
+
+- onnx: new plugin to apply ONNX neural network models to video
+
+- openaptx: aptX and aptX-HD codecs using libopenaptx (v0.2.0)
+
+- qroverlay, debugqroverlay: new elements that allows overlaying data
+ on top of video in form of a QR code
+
+- cvtracker: new OpenCV-based tracker element
+
+- av1parse, vp9parse: new parsers for AV1 and VP9 video
+
+- va: work on the new VA-API plugin implementation for
+ hardware-accelerated video decoding and encoding has continued at
+ pace, with various new decoders and filters having joined the
+ initial vah264dec:
+
+ - vah265dec: VA-API H.265 decoder
+ - vavp8dec: VA-API VP8 decoder
+ - vavp9dec: VA-API VP9 decoder
+ - vaav1dec: VA-API AV1 decoder
+ - vampeg2dec: VA-API MPEG-2 decoder
+ - vadeinterlace: : VA-API deinterlace filter
+ - vapostproc: : VA-API postproc filter (color conversion,
+ resizing, cropping, color balance, video rotation, skin tone
+ enhancement, denoise, sharpen)
+
+ See Víctor’s blog post “GstVA in GStreamer 1.20” for more details
+ and what’s coming up next.
+
+- vaapiav1dec: new AV1 decoder element (in gstreamer-vaapi)
+
+- msdkav1dec: hardware-accelerated AV1 decoder using the Intel Media
+ SDK / oneVPL
+
+- nvcodec plugin for NVIDIA NVCODEC API for hardware-accelerated video
+ encoding and decoding:
+
+ - cudaconvert, cudascale: new CUDA based video color space convert
+ and rescale elements
+ - cudaupload, cudadownload: new helper elements for memory
+ transfer between CUDA and system memory spaces
+ - nvvp8sldec, nvvp9sldec: new GstCodecs-based VP8/VP9 decoders
+
+- Various new hardware-accelerated elements for Windows:
+
+ - d3d11screencapturesrc: new desktop capture element, including a
+ GstDeviceProvider implementation to enumerate/select target
+ monitors for capture.
+ - d3d11av1dec and d3d11mpeg2dec: AV1 and MPEG-2 decoders
+ - d3d11deinterlace: deinterlacing filter
+ - d3d11compositor: video composing element
+ - see Windows section below for more details
+
+- new Rust plugins:
+
+ - audiornnoise: Removes noise from an audio stream
+ - awstranscribeparse: Parses AWS audio transcripts into timed text
+ buffers
+ - ccdetect: Detects if valid closed captions are present in a
+ closed captions stream
+ - cea608tojson: Converts CEA-608 Closed Captions to a JSON
+ representation
+ - cmafmux: CMAF fragmented MP4 muxer
+ - dashmp4mux: DASH fragmented MP4 muxer
+ - isofmp4mux: ISO fragmented MP4 muxer
+ - ebur128level: EBU R128 Loudness Level Measurement
+ - ffv1dec: FFV1 video decoder
+ - gtk4paintablesink: GTK4 video sink, which provides a
+ GdkPaintable that can be rendered in various widgets
+ - hlssink3: HTTP Live Streaming sink
+ - hrtfrender: Head-Related Transfer Function (HRTF) renderer
+ - hsvdetector: HSV colorspace detector
+ - hsvfilter: HSV colorspace filter
+ - jsongstenc: Wraps buffers containing any valid top-level JSON
+ structures into higher level JSON objects, and outputs those as
+ ndjson
+ - jsongstparse: Parses ndjson as output by jsongstenc
+ - jsontovtt: converts JSON to WebVTT subtitles
+ - regex: Applies regular expression operations on text
+ - roundedcorners: Adds rounded corners to video
+ - spotifyaudiosrc: Spotify source
+ - textahead: Display upcoming text buffers ahead (e.g. for
+ Karaoke)
+ - transcriberbin: passthrough bin that transcribes raw audio to
+ closed captions using awstranscriber and puts the captions as
+ metas onto the video
+ - tttojson: Converts timed text to a JSON representation
+ - uriplaylistbin: Playlist source bin
+ - webpdec-rs: WebP image decoder with animation support
+
+- New plugin codecalpha with elements to assist with WebM Alpha
+ decoding
+
+ - codecalphademux: Split stream with GstVideoCodecAlphaMeta into
+ two streams
+ - alphacombine: Combine two raw video stream (I420 or NV12) as one
+ stream with alpha channel (A420 or AV12)
+ - vp8alphadecodebin: A bin to handle software decoding of VP8 with
+ alpha
+ - vp9alphadecodebin: A bin to handle software decoding of VP9 with
+ alpha
+
+- New hardware accelerated elements for Linux:
+
+ - v4l2slmpeg2dec: Support for Linux Stateless MPEG2 decoders
+ - v4l2slvp9dec: Support for Linux Stateless VP9 decoders
+ - v4l2slvp8alphadecodebin: Support HW accelerated VP8 with alpha
+ layer decoding
+ - v4l2slvp9alphadecodebin: Support HW accelerated VP9 with alpha
+ layer decoding
New element features and additions
-- this section will be filled in in due course
+- assrender: handle more font mime types; better interaction with
+ matroskademux for embedded fonts
+
+- audiobuffersplit: Add support for specifying output buffer size in
+ bytes (not just duration)
+
+- audiolatency: new "samplesperbuffer" property so users can configure
+ the number of samples per buffer. The default value is 240 samples
+ which is equivalent to 5ms latency with a sample rate of 48000,
+ which might be larger than actual buffer size of audio capture
+ device.
+
+- audiomixer, audiointerleave, GstAudioAggregator: now keep a count of
+ samples that are dropped or processed as statistic and can be made
+ to post QoS messages on the bus whenever samples are dropped by
+ setting the "qos-messages" property on input pads.
+
+- audiomixer, compositor: improved handling of new inputs added at
+ runtime. New API was added to the GstAggregator base class to allow
+ subclasses to opt into an aggregation mode where inactive pads are
+ ignored when processing input buffers
+ (gst_aggregator_set_ignore_inactive_pads(),
+ gst_aggregator_pad_is_inactive()). An “inactive pad” in this context
+ is a pad which, in live mode, hasn’t yet received a first buffer,
+ but has been waited on at least once. What would happen usually in
+ this case is that the aggregator would wait for data on this pad
+ every time, up to the maximum configured latency. This would
+ inadvertently push mixer elements in live mode to the configured
+ latency envelope and delay processing when new inputs are added at
+ runtime until these inputs have actually produced data. This is
+ usually undesirable. With this new API, new inputs can be added
+ (requested) and configured and they won’t delay the data processing.
+ Applications can opt into this new behaviour by setting the
+ "ignore-inactive-pads" property on compositor, audiomixer or other
+ GstAudioAggregator-based elements.
+
+- cccombiner: implement “scheduling” of captions. So far cccombiner’s
+ behaviour was essentially that of a funnel: it strictly looked at
+ input timestamps to associate together video and caption buffers.
+ Now it will try to smoothly schedule caption buffers in order to
+ have exactly one per output video buffer. This might involve
+ rewriting input captions, for example when the input is CDP then
+ sequence counters are rewritten, time codes are dropped and
+ potentially re-injected if the input video frame had a time code
+ meta. This can also lead to the input drifting from synchronization,
+ when there isn’t enough padding in the input stream to catch up. In
+ that case the element will start dropping old caption buffers once
+ the number of buffers in its internal queue reaches a certain limit
+ (configurable via the "max-scheduled" property). The new original
+ funnel-like behaviour can be restored by setting the "scheduling"
+ property to FALSE.
+
+- ccconverter: new "cdp-mode" property to specify which sections to
+ include in CDP packets (timecode, CC data, service info). Various
+ software, including ffmpeg’s Decklink support, fails parsing CDP
+ packets that contain anything but CC data in the CDP packets.
+
+- clocksync: new "sync-to-first" property for automatic timestamp
+ offset setup: if set clocksync will set up the "ts-offset" value
+ based on the first buffer and the pipeline’s running time when the
+ first buffer arrived. The newly configured "ts-offset" in this case
+ would be the value that allows outputting the first buffer without
+ waiting on the clock. This is useful for example to feed a non-live
+ input into an already-running pipeline.
+
+- compositor:
+
+ - multi-threaded input conversion and compositing. Set the
+ "max-threads" property to activate this.
+ - new "sizing-policy" property to support display aspect ratio
+ (DAR)-aware scaling. By default the image is scaled to fill the
+ configured destination rectangle without padding and without
+ keeping the aspect ratio. With sizing-policy=keep-aspect-ratio
+ the input image is scaled to fit the destination rectangle
+ specified by GstCompositorPad:{xpos, ypos, width, height}
+ properties preserving the aspect ratio. As a result, the image
+ will be centered in the destination rectangle with padding if
+ necessary.
+ - new "zero-size-is-unscaled" property on input pads. By default
+ pad width=0 or pad height=0 mean that the stream should not be
+ scaled in that dimension. But if the "zero-size-is-unscaled"
+ property is set to FALSE a width or height of 0 is instead
+ interpreted to mean that the input image on that pad should not
+ be composited, which is useful when creating animations where an
+ input image is made smaller and smaller until it disappears.
+ - improved handling of new inputs at runtime via
+ "ignore-inactive-pads"property (see above for details)
+ - allow output format with alpha even if none of the inputs have
+ alpha (also glvideomixer and other GstVideoAggregator
+ subclasses)
+
+- dashsink: add h265 codec support and signals for allowing custom
+ playlist/fragment output
+
+- decodebin3:
+
+ - improved decoder selection, especially for hardware decoders
+ - make input activation “atomic” when adding inputs dynamically
+ - better interleave handling: take into account decoder latency
+ for interleave size
+
+- decklink:
+
+ - Updated DeckLink SDK to 11.2 to support DeckLink 8K Pro
+ - decklinkvideosrc:
+ - More accurate and stable capture timestamps: use the
+ hardware reference clock time when the frame was finished
+ being captured instead of a clock time much further down the
+ road.
+ - Automatically detect widescreen vs. normal NTSC/PAL
+
+- encodebin:
+
+ - add “smart encoding” support for H.265, VP8 and VP9 (i.e. only
+ re-encode where needed and otherwise pass through encoded video
+ as-is).
+ - H264/H265 smart encoding improvements: respect user-specified
+ stream-format, but if not specified default to avc3/hvc1 with
+ in-band SPS/PPS/VPS signalling for more flexibility.
+ - new encodebin2 element with dynamic/sometimes source pads in
+ order to support the option of doing the muxing outside of
+ encodebin, e.g. in combination with splitmuxsink.
+ - add APIs to set element properties on encoding profiles (see
+ below)
+
+- errorignore: new "ignore-eos" property to also ignore FLOW_EOS from
+ downstream elements
+
+- giosrc: add support for growing source files: applications can
+ specify that the underlying file being read is growing by setting
+ the "is-growing" property. If set, the source won’t EOS when it
+ reaches the end of the file, but will instead start monitoring it
+ and will start reading data again whenever a change is detected. The
+ new "waiting-data" and "done-waiting-data" signals keep the
+ application informed about the current state.
+
+- gtksink, gtkglsink:
+
+ - scroll event support: forwarded as navigation events into the
+ pipeline
+ - "video-aspect-ratio-override" property to force a specific
+ aspect ratio
+ - "rotate-method" property and support automatic rotation based on
+ image tags
+
+- identity: new "stats" property allows applications to retrieve the
+ number of bytes and buffers that have passed through so far.
+
+- interlace: add support for more formats, esp 10-bit, 12-bit and
+ 16-bit ones
+
+- jack: new "low-latency" property for automatic latency-optimized
+ setting and "port-names" property to select ports explicitly
+
+- jpegdec: support output conversion to RGB using libjpeg-turbo (for
+ certain input files)
+
+- line21dec:
+
+ - "mode" property to control whether and how detected closed
+ captions should be inserted in the list of existing close
+ caption metas on the input frame (if any): add, drop, or
+ replace.
+ - "ntsc-only" property to only look for captions if video has NTSC
+ resolution
+
+- line21enc: new "remove-caption-meta" to remove metas from output
+ buffers after encoding the captions into the video data; support for
+ CDP closed captions
+
+- matroskademux, matroskamux: Add support for ffv1, a lossless
+ intra-frame video coding format.
+
+- matroskamux: accept in-band SPS/PPS/VPS for H264 and H265
+ (i.e. stream-format avc3 and hev1) which allows on-the-fly
+ profile/level/resolution changes.
+
+- matroskamux: new "cluster-timestamp-offset" property, useful for use
+ cases where the container timestamps should map to some absolute
+ wall clock time, for example.
+
+- rtpsrc: add "caps" property to allow explicit setting of the caps
+ where needed
+
+- mpegts: support SCTE-35 passthrough via new "send-scte35-events"
+ property on MPEG-TS demuxer tsdemux. When enabled, SCTE 35 sections
+ (eg ad placement opportunities) are forwarded as events donwstream
+ where they can be picked up again by mpegtsmux. This required a
+ semantic change in the SCTE-35 section API: timestamps are now in
+ running time instead of muxer pts.
+
+- tsdemux: Handle PCR-less MPEG-TS streams; more robust timestamp
+ handling in certain corner cases and for poorly muxed streams.
+
+- mpegtsmux:
+
+ - More conformance improvements to make MPEG-TS analyzers happy:
+ - PCR timing accuracy: Improvements to the way mpegtsmux
+ outputs PCR observations in CBR mode, so that a PCR
+ observation is always inserted when needed, so that we never
+ miss the configured pcr-interval, as that triggers various
+ MPEG-TS analyser errors.
+ - Improved PCR/SI scheduling
+ - Don’t write PCR until PAT/PMT are output to make sure streams
+ start cleanly with a PAT/PMT.
+ - Allow overriding the automatic PMT PID selection via
+ application-supplied PMT_%d fields in the prog-map
+ structure/property.
+
+- mp4mux:
+
+ - new "first-moov-then-finalise" mode for fragmented output where
+ the output will start with a self-contained moov atom for the
+ first fragment, and then produce regular fragments. Then at the
+ end when the file is finalised, the initial moov is invalidated
+ and a new moov is written covering the entire file. This way the
+ file is a “fragmented mp4” file while it is still being written
+ out, and remains playable at all times, but at the end it is
+ turned into a regular mp4 file (with former fragment headers
+ remaining as unused junk data in the file).
+ - support H.264 avc3 and H.265 hvc1 stream formats as input where
+ the codec data is signalled in-band inside the bitstream instead
+ of caps/file headers.
+ - support profile/level/resolution changes for H264/H265 input
+ streams (i.e. codec data changing on the fly). Each codec_data
+ is put into its own SampleTableEntry inside the stsd, unless the
+ input is in avc3 stream format in which case it’s written
+ in-band an not in the headers.
+
+- multifilesink: new ""min-keyframe-distance"" property to make
+ minimum distance between keyframes in next-file=key-frame mode
+ configurable instead of hard-coding it to 10 seconds.
+
+- mxfdemux has seen a big refactoring to support non-frame wrappings
+ and more accurate timestamp/seek handling for some formats
+
+- msdk plugin for hardware-accelerated video encoding and decoding
+ using the Intel Media SDK:
+
+ - oneVPL support (Intel oneAPI Video Processing Library)
+ - AV1 decoding support
+ - H.264 decoder now supports constrained-high and progressive-high
+ profiles
+ - H.264 encoder:
+ - more configuration options (properties):
+ "intra-refresh-type", "min-qp" , "max-qp", "p-pyramid",
+ "dblk-idc"
+ - H.265 encoder:
+ - can output main-still-picture profile
+ - now inserts HDR SEIs (mastering display colour volume and
+ content light level)
+ - more configuration options (properties):
+ "intra-refresh-type", "min-qp" , "max-qp", "p-pyramid",
+ "b-pyramid", "dblk-idc", "transform-skip"
+ - support for RGB 10bit format
+ - External bitrate control in encoders
+ - Video post proc element msdkvpp gained support for 12-bit pixel
+ formats P012_LE, Y212_LE and Y412_LE
+
+- nvh264sldec: interlaced stream support
+
+- openh264enc: support main, high, constrained-high and
+ progressive-high profiles
+
+- openjpeg: support for multithreaded decoding and encoding
+
+- rtspsrc: now supports IPv6 also for tunneled mode (RTSP-over-HTTP);
+ new "ignore-x-server-reply" property to ignore the
+ x-server-ip-address server header reply in case of HTTP tunneling,
+ as it is often broken.
+
+- souphttpsrc: Runtime compatibility support for libsoup2 and
+ libsoup3. libsoup3 is the latest major version of libsoup, but
+ libsoup2 and libsoup3 can’t co-exist in the same process because
+ there is no namespacing or versioning for GObject types. As a
+ result, it would be awkward if the GStreamer souphttpsrc plugin
+ linked to a specific version of libsoup, because it would only work
+ with applications that use the same version of libsoup. To make this
+ work, the soup plugin now tries to determine the libsoup version
+ used by the application (and its other dependencies) at runtime on
+ systems where GStreamer is linked dynamically. libsoup3 support is
+ still considered somewhat experimental at this point.
+
+- srtsrc, srtsink: add signals for the application to accept/reject
+ incoming connections
+
+- timeoverlay: new elapsed-running-time time mode which shows the
+ running time since the first running time (and each flush-stop).
+
+- udpsrc: new timestamping mode to retrieve packet receive timestamps
+ from the kernel via socket control messages (SO_TIMESTAMPNS) on
+ supported platforms
+
+- uritranscodebin: new setup-source and element-setup signals for
+ applications to configure elements used
+
+- v4l2codecs plugin gained support for 4x4 and 32x32 tile formats
+ enabling some platforms or direct renders. Important memory usage
+ improvement.
+
+- v4l2slh264dec now implements the final Linux uAPI as shipped on
+ Linux 5.11 and later.
+
+- valve: add "drop-mode" property and provide two new modes of
+ operation: in drop-mode=forward-sticky-events sticky events
+ (stream-start, segment, tags, caps, etc.) are forwarded downstream
+ even when dropping is enabled; drop-mode=transform-to-gap will in
+ addition also convert buffers into gap events when dropping is
+ enabled, which lets downstream elements know that time is advancing
+ and might allow for preroll in many scenarios. By default all events
+ and all buffers are dropped when dropping is enabled, which can
+ cause problems with caps negotiation not progressing or branches not
+ prerolling when dropping is enabled.
+
+- videocrop: support for many more pixel formats, e.g. planar YUV
+ formats with > 8bits and GBR* video formats; can now also accept
+ video not backed by system memory as long as downstream supports the
+ GstCropMeta
+
+- videotestsrc: new smpte-rp-219 pattern for SMPTE75 RP-219 conformant
+ color bars
+
+- vp8enc: finish support for temporal scalability: two new properties
+ ("temporal-scalability-layer-flags",
+ "temporal-scalability-layer-sync-flags") and a unit change on the
+ "temporal-scalability-target-bitrate" property (now expects bps);
+ also make temporal scalability details available to RTP payloaders
+ as buffer metadata.
+
+- vp9enc: new properties to tweak encoder performance:
+
+ - "aq-mode" to configure adaptive quantization modes
+ - "frame-parallel-decoding" to configure whether to create a
+ bitstream that reduces decoding dependencies between frames
+ which allows staged parallel processing of more than one video
+ frames in the decoder. (Defaults to TRUE)
+ - "row-mt", "tile-columns" and "tile-rows" so multithreading can
+ be enabled on a per-tile basis, instead of on a per tile-column
+ basis. In combination with the new "tile-rows" property, this
+ allows the encoder to make much better use of the available CPU
+ power.
+
+- vp9dec, vp9enc: add support for 10-bit 4:2:0 and 4:2:2 YUV, as well
+ as 8-bit 4:4:4
+
+- vp8enc, vp9enc now default to “good quality” for the deadline
+ property rather then “best quality”. Having the deadline set to best
+ quality causes the encoder to be absurdly slow, most real-life users
+ will prefer good-enough quality with better performance instead.
+
+- wpesrc:
+
+ - implement audio support: a new sometimes source pad will be
+ created for each audio stream created by the web engine.
+ - move wpesrc to wpevideosrc and add a wrapper bin wpesrc to also
+ support audio
+ - also handles web:// URIs now (same as cefsrc)
+ - post messages with the estimated load progress on the bus
-Plugin and library moves
+- x265enc: add negative DTS support, which means timestamps are now
+ offset by 1h same as with x264enc
+
+RTP Payloaders and Depayloaders
+
+- rtpisacpay, rtpisacdepay: new RTP payloader and depayloader for iSAC
+ audio codec
+
+- rtph264depay:
+
+ - new "request-keyframe" property to make the depayloader
+ automatically request a new keyframe from the sender on packet
+ loss, consistent with the new property on rtpvp8depay.
+ - new "wait-for-keyframe" property to make depayloader wait for a
+ new keyframe at the beginning and after packet loss (only
+ effective if the depayloader outputs AUs), consistent with the
+ existing property on rtpvp8depay.
+
+- rtpopuspay, rtpopusdepay: support libwebrtc-compatible multichannel
+ audio in addition to the previously supported multichannel audio
+ modes
+
+- rtpopuspay: add DTX (Discontinuous Transmission) support
+
+- rtpvp8depay: new "request-keyframe" property to make the depayloader
+ automatically request a new keyframe from the sender on packet loss.
+
+- rtpvp8pay: temporal scaling support
-- this section will be filled in in due course
+- rtpvp9depay: Improved SVC handling (aggregate all layers)
+
+RTP Infrastructure
+
+- rtpst2022-1-fecdec, rtpst2022-1-fecenc: new elements providing SMPTE
+ 2022-1 2-D Forward Error Correction. More details in Mathieu’s blog
+ post.
+
+- rtpreddec: BUNDLE support
+
+- rtpredenc, rtpulpfecenc: add support for Transport-wide Congestion
+ Control (TWCC)
+
+- rtpsession: new "twcc-feedback-interval" property to allow RTCP TWCC
+ reports to be scheduled on a timer instead of per marker-bit.
+
+Plugin and library moves
- There were no plugin moves or library moves in this cycle.
The following elements or plugins have been removed:
-- this section will be filled in in due course
-
- The ofa audio fingerprinting plugin has been removed. The MusicIP
database has been defunct for years so this plugin is likely neither
useful nor used by anyone.
Miscellaneous API additions
-- this section will be filled in in due course
+Core
+
+- gst_buffer_new_memdup() is a convenience function for the
+ widely-used gst_buffer_new_wrapped(g_memdup(data,size),size)
+ pattern.
+
+- gst_caps_features_new_single() creates a new single GstCapsFeatures,
+ avoiding the need to use the vararg function with NULL terminator
+ for simple cases.
+
+- gst_element_type_set_skip_documentation() can be used by plugins to
+ signal that certain elements should not be included in the GStreamer
+ plugin documentation. This is useful for plugins where elements are
+ registered dynamically based on hardware capabilities and/or where
+ the available plugins and properties vary from system to system.
+ This is used in the d3d11 plugin for example to ensure that only the
+ list of default elements is advertised in the documentation.
+
+- gst_type_find_suggest_empty_simple() is a new convenience function
+ for typefinders for cases where there’s only a media type and no
+ other fields.
+
+- New API to create elements and set properties at construction time,
+ which is not only convenient, but also allows GStreamer elements to
+ have construct-only properties: gst_element_factory_make_full(),
+ gst_element_factory_make_valist(),
+ gst_element_factory_make_with_properties(),
+ gst_element_factory_create_full(),
+ gst_element_factory_create_valist(),
+ gst_element_factory_create_with_properties().
+
+- GstSharedTaskPool: new “shared” task pool subclass with slightly
+ different default behaviour than the existing GstTaskPool which
+ would create unlimited number of threads for new tasks. The shared
+ taskpool creates up to N threads (default: 1) and then distributes
+ pending tasks to those threads round-robin style, and blocks if no
+ thread is available. It is possible to join tasks. This can be used
+ by plugins to implement simple multi-threaded processing and is used
+ for the new multi-threaded video conversion and compositing done in
+ GstVideoAggregator, videoconverter and compositor.
+
+Plugins Base Utils library
+
+- GstDiscoverer:
+
+ - gst_discoverer_container_info_get_tags() was added to retrieve
+ global/container tags (vs. per-stream tags). Per-Stream tags can
+ be retrieved via the existing
+ gst_discoverer_stream_info_get_tags().
+ gst_discoverer_info_get_tags(), which for many files returns a
+ confusing mix of stream and container tags, has been deprecated
+ in favour of the container/stream-specific functions.
+ - gst_discoverer_stream_info_get_stream_number() returns a unique
+ integer identifier for a given stream within the given
+ GstDiscoverer context. (If this matches the stream number inside
+ the container bitstream that’s by coincidence and not by
+ design.)
+
+- gst_pb_utils_get_caps_description_flags() can be used to query
+ whether certain caps represent a container, audio, video, image,
+ subtitles, tags, or something else. This only works for formats
+ known to GStreamer.
+
+- gst_pb_utils_get_file_extension_from_caps() returns a possible file
+ extension for given caps.
+
+- gst_codec_utils_h264_get_profile_flags_level(): Parses profile,
+ flags, and level from H264 AvcC codec_data. The format of H264 AVCC
+ extradata/sequence_header is documented in the ITU-T H.264
+ specification section 7.3.2.1.1 as well as in ISO/IEC 14496-15
+ section 5.3.3.1.2.
+
+- gst_codec_utils_caps_get_mime_codec() to convert caps to a RFC 6381
+ compatible MIME codec string codec. Useful for providing the codecs
+ field inside the Content-Type HTTP header for containerized formats,
+ such as mp4 or matroska.
+
+GStreamer OpenGL integration library and plugins
+
+- glcolorconvert: added suppport for converting the video formats
+ A420, AV12, BGR, BGRA, RGBP and BGRP.
+
+- Added support to GstGLBuffer for persistent buffer mappings where a
+ Pixel Buffer Object (PBO) can be mapped by both the CPU and the GPU.
+ This removes a memcpy() when uploading textures or vertices
+ particularly when software decoders (e.g. libav) are direct
+ rendering into our memory. Improves transfer performance
+ significantly. Requires OpenGL 4.4, GL_ARB_buffer_storage or
+ GL_EXT_buffer_storage
+
+- Added various helper functions for handling 4x4 matrices of affine
+ transformations as used by GstVideoAffineTransformationMeta.
+
+- Add support to GstGLContext for allowing the application to control
+ the config (EGLConfig, GLXConfig, etc) used when creating the OpenGL
+ context. This allows the ability to choose between RGB16 or RGB10A2
+ or RGBA8 back/front buffer configurations that were previously
+ hardcoded. GstGLContext also supports retrieving the configuration
+ it was created with or from an externally provide OpenGL context
+ handle. This infrastructure is also used to create a compatible
+ config from an application/externally provided OpenGL context in
+ order to improve compatibility with other OpenGL frameworks and GUI
+ toolkits. A new environment variable GST_GL_CONFIG was also added to
+ be able to request a specific configuration from the command line.
+ Note: different platforms will have different functionality
+ available.
+
+- Add support for choosing between EGL and WGL at runtime when running
+ on Windows. Previously this was a build-time switch. Allows use in
+ e.g. Gtk applications on Windows that target EGL/ANGLE without
+ recompiling GStreamer. gst_gl_display_new_with_type() can be used by
+ applications to choose a specific display type to use.
+
+- Build fixes to explicitly check for Broadcom-specific libraries on
+ older versions of the Raspberry Pi platform. The Broadcom OpenGL ES
+ and EGL libraries have different filenames. Using the vc4 Mesa
+ driver on the Raspberry Pi is not affected.
+
+- Added support to glupload and gldownload for transferring RGBA
+ buffers using the memory:NVMM available on the Nvidia Tegra family
+ of embedded devices.
+
+- Added support for choosing libOpenGL and libGLX as used in a GLVND
+ environment on unix-based platforms. This allows using desktop
+ OpenGL and EGL without pulling in any GLX symbols as would be
+ required with libGL.
+
+Video library
+
+- New raw video formats:
+
+ - AV12 (NV12 with alpha plane)
+ - RGBP and BGRP (planar RGB formats)
+ - ARGB64 variants with specified endianness instead of host
+ endianness:
+ - ARGB64_LE, ARGB64_BE
+ - RGBA64_BE, RGBA64_LE
+ - BGRA64_BE, BGRA64_LE
+ - ABGR64_BE, ABGR64_LE
+
+- gst_video_orientation_from_tag() is new convenience API to parse the
+ image orientation from a GstTagList.
+
+- GstVideoDecoder subframe support (see below)
+
+- GstVideoCodecState now also carries some HDR metadata
+
+- Ancillary video data: implement transform functions for AFD/Bar
+ metas, so they will be forwarded in more cases
+
+MPEG-TS library
+
+This library only handles section parsing and such, see above for
+changes to the actual mpegtsmux and mpegtsdemux elements.
+
+- many additions and improvements to SCTE-35 section parsing
+- new API for fetching extended descriptors:
+ gst_mpegts_find_descriptor_with_extension()
+- add support for SIT sections (Selection Information Tables)
+- expose event-from-section constructor gst_event_new_mpegts_section()
+- parse Audio Preselection Descriptor needed for Dolby AC-4
+
+GstWebRTC library + webrtcbin
+
+- Change the way in which sink pads and transceivers are matched
+ together to support easier usage. If a pad is created without a
+ specific index (i.e. using sink_%u as the pad template), then an
+ available compatible transceiver will be searched for. If a specific
+ index is requested (i.e. sink_1) then if a transceiver for that
+ m-line already exists, that transceiver must match the new sink pad
+ request. If there is no transceiver available in either scenario, a
+ new transceiver is created. If a mixture of both sink_1 and sink_%u
+ requests result in an impossible situation, an error will be
+ produced at pad request time or from create offer/answer.
+
+- webrtcbin now uses regular ICE nomination instead of libnice’s
+ default of aggressive ICE nomination. Regular ICE nomination is the
+ default recommended by various relevant standards and improves
+ connectivity in specific network scenarios.
+
+- Add support for limiting the port range used for RTP with the
+ addition of the min-rtp-port and max-rtp-port properties on the ICE
+ object.
+
+- Expose the SCTP transport as a property on webrtcbin to more closely
+ match the WebRTC specification.
+
+- Added support for taking into account the data channel transport
+ state when determining the value of the "connection-state" property.
+ Previous versions of the WebRTC spec did not include the data
+ channel state when computing this value.
+
+- Add configuration for choosing the size of the underlying sockets
+ used for transporting media data
+
+- Always advertise support for the transport-cc RTCP feedback protocol
+ as rtpbin supports it. For full support, the configured caps (input
+ or through codec-preferences) need to include the relevant RTP
+ header extension.
+
+- Numerous fixes to caps and media handling to fail-fast when an
+ incompatible situation is detected.
+
+- Improved support for attaching the required media after a remote
+ offer has been set.
+
+- Add support for dynamically changing the amount of FEC used for a
+ particular stream.
+
+- webrtcbin now stops further SDP processing at the first error it
+ encounters.
+
+- Completed support for either local or the remote closing a data
+ channel.
+
+- Various fixes when performing BUNDLEing of the media streams in
+ relation to RTX and FEC usage.
+
+- Add support for writing out QoS DSCP marking on outgoing packets to
+ improve reliability in some network scenarios.
+
+- Improvements to the statistics returned by the get-stats signal
+ including the addition of the raw statistics from the internal
+ RTPSource, the TWCC stats when available.
+
+- The webrtc library does not expose any objects anymore with public
+ fields. Instead properties have been added to replace that
+ functionality. If you are accessing such fields in your application,
+ switch to the corresponding properties.
+
+GstCodecs and Video Parsers
+
+- Support for render delays to improve throughput across all CODECs
+ (used with NVDEC and V4L2).
+- lots of improvements to parsers and the codec parsing decoder base
+ classes (H264, H265, VP8, VP9, AV1, MPEG-2) used for various
+ hardware-accelerate decoder APIs.
+
+Bindings support
+
+- gst_allocation_params_new() allocates a GstAllocationParams struct
+ on the heap. This should only be used by bindings (and freed via
+ gst_allocation_params_free() then). In C code you would allocate
+ this on the stack and only init it in place.
+
+- gst_debug_log_literal() can be used to log a string to the debug log
+ without going through any printf format expansion and associated
+ overhead. This is mostly useful for bindings such as the Rust
+ bindings which may have done their own formatting already .
+
+- Provide non-inlined versions of refcounting APIs for various
+ GStreamer mini objects, so that they can be consumed by bindings
+ (e.g. gstreamer-sharp): gst_buffer_ref, gst_buffer_unref,
+ gst_clear_buffer, gst_buffer_copy, gst_buffer_replace,
+ gst_buffer_list_ref, gst_buffer_list_unref, gst_clear_buffer_list,
+ gst_buffer_list_copy, gst_buffer_list_replace, gst_buffer_list_take,
+ gst_caps_ref, gst_caps_unref, gst_clear_caps, gst_caps_replace,
+ gst_caps_take, gst_context_ref, gst_context_unref, gst_context_copy,
+ gst_context_replace, gst_event_replace, gst_event_steal,
+ gst_event_take, gst_event_ref, gst_event_unref, gst_clear_event,
+ gst_event_copy, gst_memory_ref, gst_memory_unref, gst_message_ref,
+ gst_message_unref, gst_clear_message, gst_message_copy,
+ gst_message_replace, gst_message_take, gst_promise_ref,
+ gst_promise_unref, gst_query_ref, gst_query_unref, gst_clear_query,
+ gst_query_copy, gst_query_replace, gst_query_take, gst_sample_ref,
+ gst_sample_unref, gst_sample_copy, gst_tag_list_ref,
+ gst_tag_list_unref, gst_clear_tag_list, gst_tag_list_replace,
+ gst_tag_list_take, gst_uri_copy, gst_uri_ref, gst_uri_unref,
+ gst_clear_uri.
+
+- expose a GType for GstMiniObject
+
+- gst_device_provider_probe() now returns non-floating device object
+
+API Deprecations
+
+- gst_element_get_request_pad() has been deprecated in favour of the
+ newly-added gst_element_request_pad_simple() which does the exact
+ same thing but has a less confusing name that hopefully makes clear
+ that the function request a new pad rather than just retrieves an
+ already-existing request pad.
+
+- gst_discoverer_info_get_tags(), which for many files returns a
+ confusing mix of stream and container tags, has been deprecated in
+ favour of the container-specific and stream-specific functions,
+ gst_discoverer_container_info_get_tags() and
+ gst_discoverer_stream_info_get_tags().
+
+- gst_video_sink_center_rect() was deprecated in favour of the more
+ generic newly-added gst_video_center_rect().
+
+- The GST_MEMORY_FLAG_NO_SHARE flag has been deprecated, as it tends
+ to cause problems and prevents sub-buffering. If pooling or lifetime
+ tracking is required, memories should be allocated through a custom
+ GstAllocator instead of relying on the lifetime of the buffers the
+ memories were originally attached to, which is fragile anyway.
+
+- The GstPlayer high-level playback library is being replaced with the
+ new GstPlay library (see above). GstPlayer should be considered
+ deprecated at this point and will be marked as such in the next
+ development cycle. Applications should be ported to GstPlay.
+
+- Gstreamer Editing Services: ges_video_transition_set_border(),
+ ges_video_transition_get_border()
+ ges_video_transition_set_inverted()
+ ges_video_transition_is_inverted() have been deprecated, use
+ ges_timeline_element_set_children_properties() instead.
Miscellaneous performance, latency and memory optimisations
-- this section will be filled in in due course
+More video conversion fast paths
-Miscellaneous other changes and enhancements
+- v210 ↔ I420, YV12, Y42B, UYVY and YUY2
+- A420 → RGB
+
+Less jitter when waiting on the system clock
+
+- Better system clock wait accuracy, less jitter: where available,
+ clock_nanosleep is used for higher accuracy for waits below 500
+ usecs, and waits below 2ms will first use the regular waiting system
+ and then clock_nanosleep for the remainder. The various wait
+ implementation have a latency ranging from 50 to 500+ microseconds.
+ While this is not a major issue when dealing with a low number of
+ waits per second (for ex: video), it does introduce a non-negligible
+ jitter for synchronisation of higher packet rate systems.
+
+Video decoder subframe support
-- this section will be filled in in due course
+- The GstVideoDecoder base class gained API to process input at the
+ sub-frame level. That way video decoders can start decoding slices
+ before they have received the full input frame in its entirety (to
+ the extent this is supported by the codec, of course). This helps
+ with CPU utilisation and reduces latency.
-Tracing framework and debugging improvements
+- This functionality is now being used in the OpenJPEG JPEG 2000
+ decoder, the FFmpeg H.264 decoder (in case of NAL-aligned input) and
+ the OpenMAX H.264/H.265 decoders (in case of NAL-aligned input).
-- this section will be filled in in due course
+Miscellaneous other changes and enhancements
+
+- GstDeviceMonitor no longer fails to start just because one of the
+ device providers failed to start. That could happen for example on
+ systems where the pulseaudio device provider is installed, but
+ pulseaudio isn’t actually running but ALSA is used for audio
+ instead. In the same vein the device monitor now keeps track of
+ which providers have been started (via the new
+ gst_device_provider_is_started()) and only stops actually running
+ device providers when stopping the device monitor.
+
+- On embedded systems it can be useful to create a registry that can
+ be shared and read by multiple processes running as different users.
+ It is now possible to set the new GST_REGISTRY_MODE environment
+ variable to specify the file mode for the registry file, which by
+ default is set to be only user readable/writable.
+
+- GstNetClientClock will signal lost sync in case the remote time
+ resets (e.g. because device power cycles), by emitting the “synced”
+ signal with synced=FALSE parameter, so applications can take action.
+
+- gst_value_deserialize_with_pspec() allows deserialization with a
+ hint for what the target GType should be. This allows for example
+ passing arrays of flags through the command line or
+ gst_util_set_object_arg(), eg: foo="<bar,bar+baz>".
+
+- It’s now allowed to create an empty GstVideoOverlayComposition
+ without any rectangles by passing a NULL rectangle to
+ gst_video_overlay_composition_new(). This is useful for bindings and
+ simplifies application code in some places.
+
+Tracing framework, debugging and testing improvements
+
+- New factories tracer to list loaded elements (and other plugin
+ features). This can be useful to collect a list of elements needed
+ for an application, which then in turn can be used to create a
+ tailored minimal GStreamer build that contains just the elements
+ needed and nothing else.
+- New plugin-feature-loaded tracing hook for use by tracers like the
+ new factories tracer
+
+- GstHarness: Add gst_harness_set_live() so that harnesses can be set
+ to non-live and return is-live=false in latency queries if needed.
+ Default behaviour is to always return is-live=true in latency
+ queries.
+
+- navseek: new "hold-eos" property. When enabled, the element will
+ hold back an EOS event until the next keystroke (via navigation
+ events). This can be used to keep a video sink showing the last
+ frame of a video pipeline until a key is pressed instead of tearing
+ it down immediately on EOS.
+
+- New fakeaudiosink element: mimics an audio sink and can be used for
+ testing and CI pipelines on systems where no audio system is
+ installed or running. It differs from fakesink in that it only
+ support audio caps and syncs to the clock by default like a normal
+ audio sink. It also implements the GstStreamVolume interface like
+ most audio sinks do.
+
+- New videocodectestsink element for video codec conformance testing:
+ Calculates MD5 checksums for video frames and skips any padding
+ whilst doing so. Can optionally also write back the video data with
+ padding removed into a file for easy byte-by-byte comparison with
+ reference data.
Tools
-- this section will be filled in in due course
+gst-inspect-1.0
+
+- Can sort the list of plugins by passing --sort=name as command line
+ option
+
+gst-launch-1.0
+
+- will now error out on top-level properties that don’t exist and
+ which were silently ignored before
+- On Windows the high-resolution clock is enabled now, which provides
+ better clock and timer performance on Windows (see Windows section
+ below for more details).
+
+gst-play-1.0
+
+- New --start-position command line argument to start playback from
+ the specified position
+- Audio can be muted/unmuted in interactive mode by pressing the m
+ key.
+- On Windows the high-resolution clock is enabled now (see Windows
+ section below for more details)
+
+gst-device-monitor-1.0
+
+- New --include-hidden command line argument to also show “hidden”
+ device providers
+
+ges-launch-1.0
+
+- New interactive mode that allows seeking and such. Can be disabled
+ by passing the --no-interactive argument on the command line.
+- Option to forward tags
+- Allow using an existing clip to determine the rendering format (both
+ topology and profile) via new --profile-from command line argument.
GStreamer RTSP server
-- this section will be filled in in due course
+- GstRTSPMediaFactory gained API to disable RTCP
+ (gst_rtsp_media_factory_set_enable_rtcp(), "enable-rtcp" property).
+ Previously RTCP was always allowed for all RTSP medias. With this
+ change it is possible to disable RTCP completely, no matter if the
+ client wants to do RTCP or not.
+
+- Make a mount point of / work correctly. While not allowed by the
+ RTSP 2 spec, the RTSP 1 spec is silent on this and it is used in the
+ wild. It is now possible to use / as a mount path in
+ gst-rtsp-server, e.g. rtsp://example.com/ would work with this now.
+ Note that query/fragment parts of the URI are not necessarily
+ correctly handled, and behaviour will differ between various
+ client/server implementations; so use it if you must but don’t bug
+ us if it doesn’t work with third party clients as you’d hoped.
+
+- multithreading fixes (races, refcounting issues, deadlocks)
+
+- ONVIF audio backchannel fixes
+
+- ONVIF trick mode optimisations
+
+- rtspclientsink: new "update-sdp" signal that allows updating the SDP
+ before sending it to the server via ANNOUNCE. This can be used to
+ add additional metadata to the SDP, for example. The order and
+ number of medias must not be changed, however.
GStreamer VAAPI
-- this section will be filled in in due course
+- new AV1 decoder element (vaapiav1dec)
+
+- H264 decoder: handle stereoscopic 3D video with frame packing
+ arrangement SEI messages
+
+- H265 encoder: added Screen Content Coding extensions support
+
+- H265 decoder: gained MAIN_444_12 profile support (decoded to
+ Y412_LE), and 4:2:2 12-bits support (decoded to Y212_LE)
+
+- vaapipostproc: gained BT2020 color standard support
+
+- vaapidecode: now generates caps templates dynamically at runtime in
+ order to advertise actually supported caps instead of all
+ theoretically supported caps.
+
+- GST_VAAPI_DRM_DEVICE environment variable to force a specified DRM
+ device when a DRM display is used. It is ignored when other types of
+ displays are used. By default /dev/dri/renderD128 is used for DRM
+ display.
GStreamer OMX
-- this section will be filled in in due course
+- subframe support in H.264/H.265 decoders
GStreamer Editing Services and NLE
-- this section will be filled in in due course
+- framepositioner: new "operator" property to access blending modes in
+ the compositor
+- timeline: Implement snapping to markers
+- smart-mixer: Add support for d3d11compositor and glvideomixer
+- titleclip: add "draw-shadow" child property
+- ges:// URI support to define a timeline from a description.
+- command-line-formatter
+ - Add track management to timeline description
+ - Add keyframe support
+- ges-launch-1.0:
+ - Add an interactive mode where we can seek etc…
+ - Add option to forward tags
+ - Allow using an existing clip to determine the rendering format
+ (both topology and profile) via new --profile-from command line
+ argument.
+- Fix static build
GStreamer validate
-- this section will be filled in in due course
+- report: Add a way to force backtraces on reports even if not a
+ critical issue (GST_VALIDATE_ISSUE_FLAGS_FORCE_BACKTRACE)
+- Add a flag to gst_validate_replace_variables_in_string() allow
+ defining how to resolve variables in structs
+- Add gst_validate_bin_monitor_get_scenario() to get the bin monitor
+ scenario, which is useful for applications that use Validate
+ directly.
+- Add an expected-values parameter to wait, message-type=XX allowing
+ more precise filtering of the message we are waiting for.
+- Add config file support: each test can now use a config file for the
+ given media file used to test.
+- Add support to check properties of object properties
+- scenario: Add an "action-done" signal to signal when an action is
+ done
+- scenario: Add a "run-command" action type
+- scenario: Allow forcing running action on idle from scenario file
+- scenario: Allow iterating over arrays in foreach
+- scenario: Rename ‘interlaced’ action to ‘non-blocking’
+- scenario: Add a non-blocking flag to the wait signal
GStreamer Python Bindings
-- this section will be filled in in due course
+- Fixes for Python 3.10
+- Various build fixes
+- at least one known breaking change caused by g-i annotation changes
+ (see below)
GStreamer C# Bindings
-- this section will be filled in in due course
+- Fix GstDebugGraphDetails enum
+- Updated to latests GtkSharp
+- Updated to include GStreamer 1.20 API
GStreamer Rust Bindings and Rust Plugins
-The GStreamer Rust bindings are released separately with a different
-release cadence that’s tied to gtk-rs, but the latest release has
-already been updated for the upcoming new GStreamer 1.20 API.
-
-gst-plugins-rs, the module containing GStreamer plugins written in Rust,
-has also seen lots of activity with many new elements and plugins.
-
-What follows is a list of elements and plugins available in
-gst-plugins-rs, so people don’t miss out on all those potentially useful
-elements that have no C equivalent.
-
-- FIXME: add new elements
-
-Rust audio plugins
-
-- audiornnoise: New element for audio denoising which implements the
- noise removal algorithm of the Xiph RNNoise library, in Rust
-- rsaudioecho: Port of the audioecho element from gst-plugins-good
- rsaudioloudnorm: Live audio loudness normalization element based on
- the FFmpeg af_loudnorm filter
-- claxondec: FLAC lossless audio codec decoder element based on the
- pure-Rust claxon implementation
-- csoundfilter: Audio filter that can use any filter defined via the
- Csound audio programming language
-- lewtondec: Vorbis audio decoder element based on the pure-Rust
- lewton implementation
-
-Rust video plugins
-
-- cdgdec/cdgparse: Decoder and parser for the CD+G video codec based
- on a pure-Rust CD+G implementation, used for example by karaoke CDs
-- cea608overlay: CEA-608 Closed Captions overlay element
-- cea608tott: CEA-608 Closed Captions to timed-text (e.g. VTT or SRT
- subtitles) converter
-- tttocea608: CEA-608 Closed Captions from timed-text converter
-- mccenc/mccparse: MacCaption Closed Caption format encoder and parser
-- sccenc/sccparse: Scenarist Closed Caption format encoder and parser
-- dav1dec: AV1 video decoder based on the dav1d decoder implementation
- by the VLC project
-- rav1enc: AV1 video encoder based on the fast and pure-Rust rav1e
- encoder implementation
-- rsflvdemux: Alternative to the flvdemux FLV demuxer element from
- gst-plugins-good, not feature-equivalent yet
-- rsgifenc/rspngenc: GIF/PNG encoder elements based on the pure-Rust
- implementations by the image-rs project
-
-Rust text plugins
-
-- textwrap: Element for line-wrapping timed text (e.g. subtitles) for
- better screen-fitting, including hyphenation support for some
- languages
-
-Rust network plugins
-
-- reqwesthttpsrc: HTTP(S) source element based on the Rust
- reqwest/hyper HTTP implementations and almost feature-equivalent
- with the main GStreamer HTTP source souphttpsrc
-- s3src/s3sink: Source/sink element for the Amazon S3 cloud storage
-- awstranscriber: Live audio to timed text transcription element using
- the Amazon AWS Transcribe API
-
-Generic Rust plugins
-
-- sodiumencrypter/sodiumdecrypter: Encryption/decryption element based
- on libsodium/NaCl
-- togglerecord: Recording element that allows to pause/resume
- recordings easily and considers keyframe boundaries
-- fallbackswitch/fallbacksrc: Elements for handling potentially
- failing (network) sources, restarting them on errors/timeout and
- showing a fallback stream instead
-- threadshare: Set of elements that provide alternatives for various
- existing GStreamer elements but allow to share the streaming threads
- between each other to reduce the number of threads
-- rsfilesrc/rsfilesink: File source/sink elements as replacements for
- the existing filesrc/filesink elements
+- The GStreamer Rust bindings are released separately with a different
+ release cadence that’s tied to gtk-rs, but the latest release has
+ already been updated for the upcoming new GStreamer 1.20 API (v1_20
+ feature).
+
+- gst-plugins-rs, the module containing GStreamer plugins written in
+ Rust, has also seen lots of activity with many new elements and
+ plugins. See the New Elements section above for a list of new Rust
+ elements.
Build and Dependencies
- Meson 0.59 or newer is required to build GStreamer now.
-- FIXME: this section will be filled in in due course
+- The GLib requirement has been bumped to GLib 2.56 or newer (from
+ March 2018).
+
+- The wpe plugin now requires wpe >= 2.28 and wpebackend-fdo >= 1.8
Explicit opt-in required for build of certain plugins with (A)GPL dependencies
See Building plugins with (A)GPL-licensed dependencies for more details
and a non-exhaustive list of plugins affected.
-gst-build: replaced by Monorepo
+gst-build: replaced by mono repository
-- this section will be filled in in due course
-
-- FIXME: describe + link to Monorepo FAQ
+See mono repository section above and the GStreamer mono repository FAQ.
Cerbero
on platforms where dependencies are not readily available, such as
Windows, Android, iOS and macOS.
-General improvements
+General Cerbero improvements
-- this section will be filled in in due course
+- Plugin removed: libvisual
+- New plugins: rtpmanagerbad and rist
-macOS / iOS
+macOS / iOS specific Cerbero improvements
-- this section will be filled in in due course
+- XCode 12 support
+- macOS OS release support is now future-proof, similar to iOS
+- macOS Apple Silicon (ARM64) cross-compile support has been added
+- macOS Apple Silicon (ARM64) native support is currently experimental
-Windows
+Windows specific Cerbero improvements
-- this section will be filled in in due course
+- Visual Studio 2022 support has been added
+- bootstrap is faster since it requires building fewer build-tools
+ recipes on Windows
+- package is faster due to better scheduling of recipe stages and
+ elimination of unnecessary autotools regeneration
+- The following plugins are no longer built on Windows:
+ - a52dec (another decoder is still available in libav)
+ - dvdread
+ - resindvd
Windows MSI installer
-- this section will be filled in in due course
+- no major changes
-Linux
+Linux specific Cerbero improvements
-- this section will be filled in in due course
+- Fedora, Debian OS release support is now more future-proof
+- Amazon Linux 2 support has been added
-Android
+Android specific Cerbero improvements
-- this section will be filled in in due course
+- no major changes
Platform-specific changes and improvements
Android
-- this section will be filled in in due course
+- No major changes
macOS and iOS
-- this section will be filled in in due course
+- applemedia: add ProRes support to vtenc and vtdec
Windows
-- this section will be filled in in due course
+- On Windows the high-resolution clock is enabled now in the
+ gst-launch-1.0 and gst-play-1.0 command line tools, which provides
+ better clock and timer performance on Windows, at the cost of higher
+ power consumption. By default, without the high-resolution clock
+ enabled, the timer precision on Windows is system-dependent and may
+ be as bad as 15ms which is not good enough for many multimedia
+ applications. Developers may want to do the same in their Windows
+ applications if they think it’s a good idea for their application
+ use case, and depending on the Windows version they target. This is
+ not done automatically by GStreamer because on older Windows
+ versions (pre-Windows 10) this affects a global Windows setting and
+ also there’s a power consumption vs. performance trade-off that may
+ differ from application to application.
+
+- dxgiscreencapsrc now supports resolution changes
+
+- The wasapi2 audio plugin was rewritten and now has a higher rank
+ than the old wasapi plugin since it has a number of additional
+ features such as automatic stream routing, and no
+ known-but-hard-to-fix issues. The plugin is always built if the
+ Windows 10 SDK is available now.
+
+- The wasapi device providers now detect and notify dynamic device
+ additions/removals
+
+- d3d11screencapturesrc: new desktop capture element, including
+ GstDeviceProvider implementation to enumerate/select target monitors
+ for capture.
+
+- Direct3D11/DXVA decoder now supports AV1 and MPEG2 codecs
+ (d3d11av1dec, d3d11mpeg2dec)
+
+- VP9 decoding got more reliable and stable thanks to a newly written
+ codec parser
+
+- Support for decoding interlaced H.264/AVC streams
+
+- Hardware-accelerated video deinterlacing (d3d11deinterlace) and
+ video mixing (d3d11compositor)
+
+- Video mixing with the Direct3D11 API (d3d11compositor)
+
+- MediaFoundation API based hardware encoders gained the ability to
+ receive Direct3D11 textures as an input
+
+- Seungha’s blog post “GStreamer ❤ Windows: A primer on the cool stuff
+ you’ll find in the 1.20 release” describes many of the
+ Windows-related improvements in more detail
Linux
-- this section will be filled in in due course
+- bluez: LDAC Bluetooth audio codec support in a2dpsink and avdtpsink,
+ as well as an LDAC RTP payloader (rtpldacpay) and an LDAC audio
+ encoder (ldacenc)
+
+- kmssink: gained support for NV24, NV61, RGB16/BGR16 formats;
+ auto-detect NVIDIA Tegra driver
Documentation improvements
-- this section will be filled in in due course
+- hardware-accelerated GPU plugins will now no longer always list all
+ the element variants for all available GPUs, since those are
+ system-dependent and it’s confusing for users to see those in the
+ documentation just because the GStreamer developer who generated the
+ docs had multiple GPUs to play with at the time. Instead just show
+ the default elements.
-Possibly Breaking Changes
+Possibly Breaking and Other Noteworthy Behavioural Changes
+
+- gst_parse_launch(), gst_parse_bin_from_description() and friends
+ will now error out when setting properties that don’t exist on
+ top-level bins. They were silently ignored before.
+
+- The GstWebRTC library does not expose any objects anymore with
+ public fields. Instead properties have been added to replace that
+ functionality. If you are accessing such fields in your application,
+ switch to the corresponding properties.
-- FIXME: this section will be filled in in due course
-- MPEG-TS SCTE-35 API changes (FIXME: flesh out)
-- gst_parse_launch() and friends now error out on non-existing
- properties on top-level bins where they would silently fail and
- ignore those before.
- playbin and uridecodebin now emit the source-setup signal before the
element is added to the bin and linked so that the source element is
already configured before any scheduling query comes in, which is
- useful for elements such as appsrc or giostreamsrc. (Merge Request)
+ useful for elements such as appsrc or giostreamsrc.
+
+- The source element inside urisourcebin (used inside uridecodebin3
+ which is used inside playbin3) is no longer called "source". This
+ shouldn’t affect anyone hopefully, because there’s a "setup-source"
+ signal to configure the source element and no one should rely on
+ names of internal elements anyway.
+
+- The vp8enc element now expects bps (bits per second) for the
+ "temporal-scalability-target-bitrate" property, which is consistent
+ with the "target-bitrate" property. Since additional configuration
+ is required with modern libvpx to make temporal scaling work anyway,
+ chances are that very few people will have been using this property
+
+- vp8enc and vp9enc now default to “good quality” for the "deadline"
+ property rather then “best quality”. Having the deadline set to best
+ quality causes the encoder to be absurdly slow, most real-life users
+ will want the good quality tradeoff instead.
+
+- The experimental GstTranscoder library API in gst-plugins-bad was
+ changed from a GObject signal-based notification mechanism to a
+ GstBus/message-based mechanism akin to GstPlayer/GstPlay.
+
+- MPEG-TS SCTE-35 API: semantic change for SCTE-35 splice commands:
+ timestamps passed by the application should be in running time now,
+ since users of the API can’t really be expected to predict the local
+ PTS of the muxer.
+
+- The GstContext used by souphttpsrc to share the session between
+ multiple element instances has changed. Previously it provided
+ direct access to the internal SoupSession object, now it only
+ provides access to an opaque, internal type. This change is
+ necessary because SoupSession is not thread-safe at all and can’t be
+ shared safely between arbitrary external code and souphttpsrc.
+
+- Python bindings: GObject-introspection related Annotation fixes have
+ led to a case of a GstVideo.VideoInfo-related function signature
+ changing in the Python bindings (possibly one or two other cases
+ too). This is for a function that should never have been exposed in
+ the first place though, so the bindings are being updated to throw
+ an exception in that case, and the correct replacement API has been
+ added in form of an override.
Known Issues
-- this section will be filled in in due course
-
-- There are a couple of known WebRTC-related regressions/blockers:
-
- - webrtc: DTLS setup with Chrome is broken
- - webrtcbin: First keyframe is usually lost
+- nothing in particular at this point (but also see possibly breaking
+ changes section above)
Contributors
-- this section will be filled in in due course
+Aaron Boxer, Adam Leppky, Adam Williamson, Alba Mendez, Alejandro
+González, Aleksandr Slobodeniuk, Alexander Vandenbulcke, Alex Ashley,
+Alicia Boya García, Andika Triwidada, Andoni Morales Alastruey, Andrew
+Wesie, Andrey Moiseev, Antonio Ospite, Antonio Rojas, Arthur Crippa
+Búrigo, Arun Raghavan, Ashley Brighthope, Axel Kellermann, Baek, Bastien
+Nocera, Bastien Reboulet, Benjamin Gaignard, Bing Song, Binh Truong,
+Biswapriyo Nath, Brad Hards, Brad Smith, Brady J. Garvin, Branko
+Subasic, Camilo Celis Guzman, Chris Bass, ChrisDuncanAnyvision, Chris
+White, Corentin Damman, Daniel Almeida, Daniel Knobe, Daniel Stone,
+david, David Fernandez, David Keijser, David Phung, Devarsh Thakkar,
+Dinesh Manajipet, Dmitry Samoylov, Dmitry Shusharin, Dominique Martinet,
+Doug Nazar, Ederson de Souza, Edward Hervey, Emmanuel Gil Peyrot,
+Enrique Ocaña González, Ezequiel Garcia, Fabian Orccon, Fabrice
+Fontaine, Fernando Jimenez Moreno, Florian Karydes, Francisco Javier
+Velázquez-García, François Laignel, Frederich Munch, Fredrik Pålsson,
+George Kiagiadakis, Georg Lippitsch, Göran Jönsson, Guido Günther,
+Guillaume Desmottes, Guiqin Zou, Haakon Sporsheim, Haelwenn (lanodan)
+Monnier, Haihao Xiang, Haihua Hu, Havard Graff, He Junyan, Helmut
+Januschka, Henry Wilkes, Hosang Lee, Hou Qi, Ignacio Casal Quinteiro,
+Igor Kovalenko, Ilya Kreymer, Imanol Fernandez, Jacek Tomaszewski, Jade
+Macho, Jakub Adam, Jakub Janků, Jan Alexander Steffens (heftig), Jan
+Schmidt, Jason Carrete, Jason Pereira, Jay Douglass, Jeongki Kim, Jérôme
+Laheurte, Jimmi Holst Christensen, Johan Sternerup, John Hassell, John
+Lindgren, John-Mark Bell, Jonathan Matthew, Jordan Petridis, Jose
+Quaresma, Julian Bouzas, Julien, Kai Uwe Broulik, Kasper Steensig
+Jensen, Kellermann Axel, Kevin Song, Khem Raj, Knut Inge Hvidsten, Knut
+Saastad, Kristofer Björkström, Lars Lundqvist, Lawrence Troup, Lim Siew
+Hoon, Lucas Stach, Ludvig Rappe, Luis Paulo Fernandes de Barros, Luke
+Yelavich, Mads Buvik Sandvei, Marc Leeman, Marco Felsch, Marek Vasut,
+Marian Cichy, Marijn Suijten, Marius Vlad, Markus Ebner, Mart Raudsepp,
+Matej Knopp, Mathieu Duponchelle, Matthew Waters, Matthieu De Beule,
+Mengkejiergeli Ba, Michael de Gans, Michael Olbrich, Michael Tretter,
+Michal Dzik, Miguel Paris, Mikhail Fludkov, mkba, Nazar Mokrynskyi,
+Nicholas Jackson, Nicola Murino, Nicolas Dufresne, Niklas Hambüchen,
+Nikolay Sivov, Nirbheek Chauhan, Olivier Blin, Olivier Crete, Olivier
+Crête, Paul Goulpié, Per Förlin, Peter Boba, P H, Philippe Normand,
+Philipp Zabel, Pieter Willem Jordaan, Piotrek Brzeziński, Rafał
+Dzięgiel, Rafostar, raghavendra, Raghavendra, Raju Babannavar, Raleigh
+Littles III, Randy Li, Randy Li (ayaka), Ratchanan Srirattanamet, Raul
+Tambre, reed.lawrence, Ricky Tang, Robert Rosengren, Robert Swain, Robin
+Burchell, Roman Sivriver, R S Nikhil Krishna, Ruben Gonzalez, Ruslan
+Khamidullin, Sanchayan Maity, Scott Moreau, Sebastian Dröge, Sergei
+Kovalev, Seungha Yang, Sid Sethupathi, sohwan.park, Sonny Piers, Staz M,
+Stefan Brüns, Stéphane Cerveau, Stephan Hesse, Stian Selnes, Stirling
+Westrup, Théo MAILLART, Thibault Saunier, Tim, Timo Wischer, Tim-Philipp
+Müller, Tim Schneider, Tobias Ronge, Tom Schoonjans, Tulio Beloqui,
+tyler-aicradle, U. Artie Eoff, Ung, Val Doroshchuk, VaL Doroshchuk,
+Víctor Manuel Jáquez Leal, Vivek R, Vivia Nikolaidou, Vivienne
+Watermeier, Vladimir Menshakov, Will Miller, Wim Taymans, Xabier
+Rodriguez Calvar, Xavier Claessens, Xℹ Ruoyao, Yacine Bandou, Yinhang
+Liu, youngh.lee, youngsoo.lee, yychao, Zebediah Figura, Zhang yuankun,
+Zhang Yuankun, Zhao, Zhao Zhili, , Aleksandar Topic, Antonio Ospite,
+Bastien Nocera, Benjamin Gaignard, Brad Hards, Carlos Falgueras García,
+Célestin Marot, Corentin Damman, Corentin Noël, Daniel Almeida, Daniel
+Knobe, Danny Smith, Dave Piché, Dmitry Osipenko, Fabrice Fontaine,
+fjmax, Florian Zwoch, Guillaume Desmottes, Haihua Hu, Heinrich Kruger,
+He Junyan, Jakub Adam, James Cowgill, Jan Alexander Steffens (heftig),
+Jean Felder, Jeongki Kim, Jiri Uncovsky, Joe Todd, Jordan Petridis,
+Krystian Wojtas, Marc-André Lureau, Marcin Kolny, Marc Leeman, Mark
+Nauwelaerts, Martin Reboredo, Mathieu Duponchelle, Matthew Waters,
+Mengkejiergeli Ba, Michael Gruner, Nicolas Dufresne, Nirbheek Chauhan,
+Olivier Crête, Philippe Normand, Rafał Dzięgiel, Ralf Sippl, Robert
+Mader, Sanchayan Maity, Sangchul Lee, Sebastian Dröge, Seungha Yang,
+Stéphane Cerveau, Teh Yule Kim, Thibault Saunier, Thomas Klausner, Timo
+Wischer, Tim-Philipp Müller, Tobias Reineke, Tomasz Andrzejak, Trung Do,
+Tyler Compton, Ung, Víctor Manuel Jáquez Leal, Vivia Nikolaidou, Wim
+Taymans, wngecn, Wonchul Lee, wuchang li, Xavier Claessens, Xi Ruoyao,
+Yoshiharu Hirose, Zhao,
… and many others who have contributed bug reports, translations, sent
suggestions or helped testing.
1.20.0
-1.20.0 is scheduled to be released around October/November 2021.
+1.20.0 is scheduled to be released around early February 2022.
Schedule for 1.22
unstable development version leading up to the stable 1.22 release. The
development of 1.21/1.22 will happen in the git main branch.
-The plan for the 1.22 development cycle is yet to be confirmed.
+The plan for the 1.22 development cycle is yet to be confirmed. Assuming
+no major project-wide reorganisations in the 1.22 cycle we might try and
+aim for a release around August 2022.
1.22 will be backwards-compatible to the stable 1.20, 1.18, 1.16, 1.14,
1.12, 1.10, 1.8, 1.6, 1.4, 1.2 and 1.0 release series.
------------------------------------------------------------------------
These release notes have been prepared by Tim-Philipp Müller with
-contributions from …
+contributions from Matthew Waters, Nicolas Dufresne, Nirbheek Chauhan,
+Sebastian Dröge and Seungha Yang.
License: CC BY-SA 4.0
-This is GStreamer core 1.19.3.
+This is GStreamer core 1.19.90.
GStreamer 1.19 is the development branch leading up to the next major
stable version which will be 1.20.
<release>
<Version>
+ <revision>1.19.90</revision>
+ <branch>main</branch>
+ <name></name>
+ <created>2022-01-28</created>
+ <file-release rdf:resource="https://gstreamer.freedesktop.org/src/gstreamer/gstreamer-1.19.90.tar.xz" />
+ </Version>
+ </release>
+
+ <release>
+ <Version>
<revision>1.19.3</revision>
<branch>main</branch>
<name></name>
project('gstreamer', 'c',
- version : '1.19.3.1',
+ version : '1.19.90',
meson_version : '>= 0.59',
default_options : [ 'warning_level=1',
'buildtype=debugoptimized' ])
projects / platform / upstream / gstreamer.git / commitdiff