+GStreamer 1.18 Release Notes
+GStreamer 1.18.0 was originally released on 7 September 2020.
-GSTREAMER 1.14 RELEASE NOTES
+See https://gstreamer.freedesktop.org/releases/1.18/ for the latest
+version of this document.
+
+Last updated: Monday 7 September 2020, 10:30 UTC (log)
+Introduction
The GStreamer team is proud to announce a new major feature release in
the stable 1.x API series of your favourite cross-platform multimedia
framework!
-As always, this release is again packed with new features, bug fixes and
-other improvements.
+As always, this release is again packed with many new features, bug
+fixes and other improvements.
-GStreamer 1.14.0 was released on 19 March 2018.
-
-See https://gstreamer.freedesktop.org/releases/1.14/ for the latest
-version of this document.
+Highlights
-_Last updated: Monday 19 March 2018, 12:00 UTC (log)_
+- GstTranscoder: new high level API for applications to transcode
+ media files from one format to another
+- High Dynamic Range (HDR) video information representation and
+ signalling enhancements
-Introduction
+- Instant playback rate change support
-The GStreamer team is proud to announce a new major feature release in
-the stable 1.x API series of your favourite cross-platform multimedia
-framework!
+- Active Format Description (AFD) and Bar Data support
-As always, this release is again packed with new features, bug fixes and
-other improvements.
+- ONVIF trick modes support in both GStreamer RTSP server and client
+- Hardware-accelerated video decoding on Windows via DXVA2 /
+ Direct3D11
-Highlights
+- Microsoft Media Foundation plugin for video capture and
+ hardware-accelerated video encoding on Windows
-- WebRTC support: real-time audio/video streaming to and from web
- browsers
+- qmlgloverlay: New overlay element that renders a QtQuick scene over
+ the top of an input video stream
-- Experimental support for the next-gen royalty-free AV1 video codec
+- New imagesequencesrc element to easily create a video stream from a
+ sequence of jpeg or png images
-- Video4Linux: encoding support, stable element names and faster
- device probing
+- dashsink: Add new sink to produce DASH content
-- Support for the Secure Reliable Transport (SRT) video streaming
- protocol
+- dvbsubenc: DVB Subtitle encoder element
-- RTP Forward Error Correction (FEC) support (ULPFEC)
+- TV broadcast compliant MPEG-TS muxing with constant bitrate muxing
+ and SCTE-35 support
-- RTSP 2.0 support in rtspsrc and gst-rtsp-server
+- rtmp2: new RTMP client source and sink element implementation
-- ONVIF audio backchannel support in gst-rtsp-server and rtspsrc
+- svthevcenc: new SVT-HEVC-based H.265 video encoder
-- playbin3 gapless playback and pre-buffering support
+- vaapioverlay compositor element using VA-API
-- tee, our stream splitter/duplication element, now does allocation
- query aggregation which is important for efficient data handling and
- zero-copy
+- rtpmanager support for Google’s Transport-Wide Congestion Control
+ (twcc) RTP extension
-- QuickTime muxer has a new prefill recording mode that allows file
- import in Adobe Premiere and FinalCut Pro while the file is still
- being written.
+- splitmuxsink and splitmuxsrc gained support for auxiliary video
+ streams
-- rtpjitterbuffer fast-start mode and timestamp offset adjustment
- smoothing
+- webrtcbin now contains some initial support for renegotiation
+ involving stream addition and removal
-- souphttpsrc connection sharing, which allows for connection reuse,
- cookie sharing, etc.
+- New RTP source and sink elements to easily set up RTP streaming via
+ rtp:// URIs
-- nvdec: new plugin for hardware-accelerated video decoding using the
- NVIDIA NVDEC API
+- New Audio Video Transport Protocol (AVTP) plugin for Time-Sensitive
+ Applications
-- Adaptive DASH trick play support
+- Support for the Video Services Forum’s Reliable Internet Stream
+ Transport (RIST) TR-06-1 Simple Profile
-- ipcpipeline: new plugin that allows splitting a pipeline across
- multiple processes
+- Universal Windows Platform (UWP) support
-- Major gobject-introspection annotation improvements for large parts
- of the library API
+- rpicamsrc element for capturing from the Raspberry Pi camera
-- GStreamer C# bindings have been revived and seen many updates and
- fixes
+- RTSP Server TCP interleaved backpressure handling improvements as
+ well as support for Scale/Speed headers
-- The externally maintained GStreamer Rust bindings had many usability
- improvements and cover most of the API now. Coinciding with the 1.14
- release, a new release with the 1.14 API additions is happening.
+- GStreamer Editing Services gained support for nested timelines,
+ per-clip speed rate control and the OpenTimelineIO format.
+- Autotools build system has been removed in favour of Meson
Major new features and changes
-WebRTC support
-
-There is now basic support for WebRTC in GStreamer in form of a new
-webrtcbin element and a webrtc support library. This allows you to build
-applications that set up connections with and stream to and from other
-WebRTC peers, whilst leveraging all of the usual GStreamer features such
-as hardware-accelerated encoding and decoding, OpenGL integration,
-zero-copy and embedded platform support. And it's easy to build and
-integrate into your application too!
-
-WebRTC enables real-time communication of audio, video and data with web
-browsers and native apps, and it is supported or about to be support by
-recent versions of all major browsers and operating systems.
-
-GStreamer's new WebRTC implementation uses libnice for Interactive
-Connectivity Establishment (ICE) to figure out the best way to
-communicate with other peers, punch holes into firewalls, and traverse
-NATs.
-
-The implementation is not complete, but all the basics are there, and
-the code sticks fairly close to the PeerConnection API. Where
-functionality is missing it should be fairly obvious where it needs to
-go.
-
-For more details, background and example code, check out Nirbheek's blog
-post _GStreamer has grown a WebRTC implementation_, as well as Matthew's
-_GStreamer WebRTC_ talk from last year's GStreamer Conference in Prague.
-
-New Elements
-
-- webrtcbin handles the transport aspects of webrtc connections (see
- WebRTC section above for more details)
-
-- New srtsink and srtsrc elements for the Secure Reliable Transport
- (SRT) video streaming protocol, which aims to be easy to use whilst
- striking a new balance between reliability and latency for low
- latency video streaming use cases. More details about SRT and the
- implementation in GStreamer in Olivier's blog post _SRT in
- GStreamer_.
-
-- av1enc and av1dec elements providing experimental support for the
- next-generation royalty free video AV1 codec, alongside Matroska
- support for it.
-
-- hlssink2 is a rewrite of the existing hlssink element, but unlike
- its predecessor hlssink2 takes elementary streams as input and
- handles the muxing to MPEG-TS internally. It also leverages
- splitmuxsink internally to do the splitting. This allows more
- control over the chunk splitting and sizing process and relies less
- on the co-operation of an upstream muxer. Different to the old
- hlssink it also works with pre-encoded streams and does not require
- close interaction with an upstream encoder element.
-
-- audiolatency is a new element for measuring audio latency end-to-end
- and is useful to measure roundtrip latency including both the
- GStreamer-internal latency as well as latency added by external
- components or circuits.
-
-- 'fakevideosink is basically a null sink for video data and very
- similar to fakesink, only that it will answer allocation queries and
- will advertise support for various video-specific things such
- GstVideoMeta, GstVideoCropMeta and GstVideoOverlayCompositionMeta
- like a normal video sink would. This is useful for throughput
- testing and testing the zero-copy path when creating a new pipeline.
-
-- ipcpipeline: new plugin that allows the splitting of a pipeline into
- multiple processes. Usually a GStreamer pipeline runs in a single
- process and parallelism is achieved by distributing workloads using
- multiple threads. This means that all elements in the pipeline have
- access to all the other elements' memory space however, including
- that of any libraries used. For security reasons one might therefore
- want to put sensitive parts of a pipeline such as DRM and decryption
- handling into a separate process to isolate it from the rest of the
- pipeline. This can now be achieved with the new ipcpipeline plugin.
- Check out George's blog post _ipcpipeline: Splitting a GStreamer
- pipeline into multiple processes_ or his lightning talk from last
- year's GStreamer Conference in Prague for all the gory details.
-
-- proxysink and proxysrc are new elements to pass data from one
- pipeline to another within the same process, very similar to the
- existing inter elements, but not limited to raw audio and video
- data. These new proxy elements are very special in how they work
- under the hood, which makes them extremely powerful, but also
- dangerous if not used with care. The reason for this is that it's
- not just data that's passed from sink to src, but these elements
- basically establish a two-way wormhole that passes through queries
- and events in both directions, which means caps negotiation and
- allocation query driven zero-copy can work through this wormhole.
- There are scheduling considerations as well: proxysink forwards
- everything into the proxysrc pipeline directly from the proxysink
- streaming thread. There is a queue element inside proxysrc to
- decouple the source thread from the sink thread, but that queue is
- not unlimited, so it is entirely possible that the proxysink
- pipeline thread gets stuck in the proxysrc pipeline, e.g. when that
- pipeline is paused or stops consuming data for some other reason.
- This means that one should always shut down down the proxysrc
- pipeline before shutting down the proxysink pipeline, for example.
- Or at least take care when shutting down pipelines. Usually this is
- not a problem though, especially not in live pipelines. For more
- information see Nirbheek's blog post _Decoupling GStreamer
- Pipelines_, and also check out out the new ipcpipeline plugin for
- sending data from one process to another process (see above).
-
-- lcms is a new LCMS-based ICC color profile correction element
-
-- openmptdec is a new OpenMPT-based decoder for module music formats,
- such as S3M, MOD, XM, IT. It is built on top of a new
- GstNonstreamAudioDecoder base class which aims to unify handling of
- files which do not operate a streaming model. The wildmidi plugin
- has also been revived and is also implemented on top of this new
- base class.
-
-- The curl plugin has gained a new curlhttpsrc element, which is
- useful for testing HTTP protocol version 2.0 amongst other things.
-
-- The msdk plugin has gained a MPEG-2 video decoder(msdkmpeg2dec), VP8
- decoder(msdkvp8dec) and a VC1/WMV decoder(msdkvc1dec)
-
-Noteworthy new API
-
-- GstPromise provides future/promise-like functionality. This is used
- in the GStreamer WebRTC implementation.
-
-- GstReferenceTimestampMeta is a new meta that allows you to attach
- additional reference timestamps to a buffer. These timestamps don't
- have to relate to the pipeline clock in any way. Examples of this
- could be an NTP timestamp when the media was captured, a frame
- counter on the capture side or the (local) UNIX timestamp when the
- media was captured. The decklink elements make use of this.
-
-- GstVideoRegionOfInterestMeta: it's now possible to attach generic
- free-form element-specific parameters to a region of interest meta,
- for example to tell a downstream encoder to use certain codec
- parameters for a certain region.
-
-- gst_bus_get_pollfd can be used to obtain a file descriptor for the
- bus that can be poll()-ed on for new messages. This is useful for
- integration with non-GLib event loops.
-
-- gst_get_main_executable_path() can be used by wrapper plugins that
- need to find things in the directory where the application
- executable is located. In the same vein,
- GST_PLUGIN_DEPENDENCY_FLAG_PATHS_ARE_RELATIVE_TO_EXE can be used to
- signal that plugin dependency paths are relative to the main
- executable.
-
-- pad templates can be told about the GType of the pad subclass of the
- pad via newly-added GstPadTemplate API API or the
- gst_element_class_add_static_pad_template_with_gtype() convenience
- function. gst-inspect-1.0 will use this information to print pad
- properties.
-
-- new convenience functions to iterate over element pads without using
- the GstIterator API: gst_element_foreach_pad(),
- gst_element_foreach_src_pad(), and gst_element_foreach_sink_pad().
-
-- GstBaseSrc and appsrc have gained support for buffer lists:
- GstBaseSrc subclasses can use gst_base_src_submit_buffer_list(), and
- applications can use gst_app_src_push_buffer_list() to push a buffer
- list into appsrc.
-
-- The GstHarness unit test harness has a couple of new convenience
- functions to retrieve all pending data in the harness in form of a
- single chunk of memory.
-
-- GstAudioStreamAlign is a new helper object for audio elements that
- handles discontinuity detection and sample alignment. It will align
- samples after the previous buffer's samples, but keep track of the
- divergence between buffer timestamps and sample position (jitter).
- If it exceeds a configurable threshold the alignment will be reset.
- This simply factors out code that was duplicated in a number of
- elements into a common helper API.
-
-- The GstVideoEncoder base class implements Quality of Service (QoS)
- now. This is disabled by default and must be opted in by setting the
- "qos" property, which will make the base class gather statistics
- about the real-time performance of the pipeline from downstream
- elements (usually sinks that sync the pipeline clock). Subclasses
- can then make use of this by checking whether input frames are late
- already using gst_video_encoder_get_max_encode_time() If late, they
- can just drop them and skip encoding in the hope that the pipeline
- will catch up.
-
-- The GstVideoOverlay interface gained a few helper functions for
- installing and handling a "render-rectangle" property on elements
- that implement this interface, so that this functionality can also
- be used from the command line for testing and debugging purposes.
- The property wasn't added to the interface itself as that would
- require all implementors to provide it which would not be
- backwards-compatible.
-
-- A new base class, GstNonstreamAudioDecoder for non-stream audio
- decoders was added to gst-plugins-bad. This base-class is meant to
- be used for audio decoders that require the whole stream to be
- loaded first before decoding can start. Examples of this are module
- formats (MOD/S3M/XM/IT/etc), C64 SID tunes, video console music
- files (GYM/VGM/etc), MIDI files and others. The new openmptdec
- element is based on this.
-
-- Full list of API new in 1.14:
-- GStreamer core API new in 1.14
-- GStreamer base library API new in 1.14
-- gst-plugins-base libraries API new in 1.14
-- gst-plugins-bad: no list, mostly GstWebRTC library and new
- non-stream audio decoder base class.
-
-New RTP features and improvements
-
-- rtpulpfecenc and rtpulpfecdec are new elements that implement
- Generic Forward Error Correction (FEC) using Uneven Level Protection
- (ULP) as described in RFC 5109. This can be used to protect against
- certain types of (non-bursty) packet loss, and important packets
- such as those containing codec configuration data or key frames can
- be protected with higher redundancy. Equally, packets that are not
- particularly important can be given low priority or not be protected
- at all. If packets are lost, the receiver can then hopefully restore
- the lost packet(s) from the surrounding packets which were received.
- This is an alternative to, or rather complementary to, dealing with
- packet loss using _retransmission (rtx)_. GStreamer has had
- retransmission support for a long time, but Forward Error Correction
- allows for different trade-offs: The advantage of Forward Error
- Correction is that it doesn't add latency, whereas retransmission
- requires at least one more roundtrip to request and hopefully
- receive lost packets; Forward Error Correction increases the
- required bandwidth however, even in situations where there is no
- packet loss at all, so one will typically want to fine-tune the
- overhead and mechanisms used based on the characteristics of the
- link at the time.
-
-- New _Redundant Audio Data (RED)_ encoders and decoders for RTP as
- per RFC 2198 are also provided (rtpredenc and rtpreddec), mostly for
- chrome webrtc compatibility, as chrome will wrap ULPFEC-protected
- streams in RED packets, and such streams need to be wrapped and
- unwrapped in order to use ULPFEC with chrome.
-
-- a few new buffer flags for FEC support:
- GST_BUFFER_FLAG_NON_DROPPABLE can be used to mark important buffers,
- e.g. to flag RTP packets carrying keyframes or codec setup data for
- RTP Forward Error Correction purposes, or to prevent still video
- frames from being dropped by elements due to QoS. There already is a
- GST_BUFFER_FLAG_DROPPABLE. GST_RTP_BUFFER_FLAG_REDUNDANT is used to
- signal internally that a packet represents a redundant RTP packet
- and used in rtpstorage to hold back the packet and use it only for
- recovery from packet loss. Further work is still needed in
- payloaders to make use of these.
-
-- rtpbin now has an option for increasing timestamp offsets gradually:
- Sudden large changes to the internal ts_offset may cause timestamps
- to move backwards and may also cause visible glitches in media
- playback. The new "max-ts-offset-adjustment" and "max-ts-offset"
- properties let the application control the rate to apply changes to
- ts_offset. There have also been some EOS/BYE handling improvements
- in rtpbin.
-
-- rtpjitterbuffer has a new fast start mode: in many scenarios the
- jitter buffer will have to wait for the full configured latency
- before it can start outputting packets. The reason for that is that
- it often can't know what the sequence number of the first expected
- RTP packet is, so it can't know whether a packet earlier than the
- earliest packet received will still arrive in future. This behaviour
- can now be bypassed by setting the "faststart-min-packets" property
- to the number of consecutive packets needed to start, and the jitter
- buffer will start output packets as soon as it has N consecutive
- packets queued internally. This is particularly useful to get a
- first video frame decoded and rendered as quickly as possible.
-
-- rtpL8pay and rtpL8depay provide RTP payloading and depayloading for
- 8-bit raw audio
-
-New element features
-
-- playbin3 has gained support or gapless playback via the
- "about-to-finish" signal where users can set the uri for the next
- item to play. For non-live streams this will be emitted as soon as
- the first uri has finished downloading, so with sufficiently large
- buffers it is now possible to pre-buffer the next item well ahead of
- time (unlike playbin where there would not be a lot of time between
- "about-to-finish" emission and the end of the stream). If the stream
- format of the next stream is the same as that of the previous
- stream, the data will be concatenated via the concat element.
- Whether this will result in true gaplessness depends on the
- container format and codecs used, there might still be codec-related
- gaps between streams with some codecs.
-
-- tee now does allocation query aggregation, which is important for
- zero-copy and efficient data handling, especially for video. Those
- who want to drop allocation queries on purpose can use the identity
- element's new "drop-allocation" property for that instead.
-
-- audioconvert now has a "mix-matrix" property, which obsoletes the
- audiomixmatrix element. There's also mix matrix support in the audio
- conversion and channel mixing API.
-
-- x264enc: new "insert-vui" property to disable VUI (Video Usability
- Information) parameter insertion into the stream, which allows
- creation of streams that are compatible with certain legacy hardware
- decoders that will refuse to decode in certain combinations of
- resolution and VUI parameters; the max. allowed number of B-frames
- was also increased from 4 to 16.
-
-- dvdlpcmdec: has gained support for Blu-Ray audio LPCM.
-
-- appsrc has gained support for buffer lists (see above) and also seen
- some other performance improvements.
-
-- flvmux has been ported to the GstAggregator base class which means
- it can work in defined-latency mode with live input sources and
+Noteworthy new features and API
+
+Instant playback rate changes
+
+Changing the playback rate as quickly as possible so far always required
+a flushing seek. This generally works, but has the disadvantage of
+flushing all data from the playback pipeline and requiring the demuxer
+or parser to do a full-blown seek including resetting its internal state
+and resetting the position of the data source. It might also require
+considerable decoding effort to get to the right position to resume
+playback from at the higher rate.
+
+This release adds a new mechanism to achieve quasi-instant rate changes
+in certain playback pipelines without interrupting the flow of data in
+the pipeline. This is activated by sending a seek with the
+GST_SEEK_FLAG_INSTANT_RATE_CHANGE flag and start_type = stop_type =
+GST_SEEK_TYPE_NONE. This flag does not work for all pipelines, in which
+case it is necessary to fall back to sending a full flushing seek to
+change the playback rate. When using this flag, the seek event is only
+allowed to change the current rate and can modify the trickmode flags
+(e.g. keyframe only or not), but it is not possible to change the
+current playback position, playback direction or do a flush.
+
+This is particularly useful for streaming use cases like HLS or DASH
+where the streaming download should not be interrupted when changing
+rate.
+
+Instant rate changing is handled in the pipeline in a specific sequence
+which is detailed in the seeking design docs. Most elements don’t need
+to worry about this, only elements that sync to the clock need some
+special handling which is implemented in the GstBaseSink base class, so
+should be taken care of automatically in most normal playback pipelines
+and sink elements.
+
+See Jan’s GStreamer Conference 2019 talk “Changing Playback Rate
+Instantly” for more information.
+
+You can try this feature by passing the -i command line option to
+gst-play-1.0. It is supported at least by qtdemux, tsdemux, hlsdemux,
+and dashdemux.
+
+Google Transport-Wide Congestion Control
+
+rtpmanager now supports the parsing and generating of RTCP messages for
+the Google Transport-Wide Congestion Control RTP Extension, as described
+in:
+https://tools.ietf.org/html/draft-holmer-rmcat-transport-wide-cc-extensions-01.
+
+This “just” provides the required plumbing/infrastructure, it does not
+actually make effect any actual congestion control on the sender side,
+but rather provides information for applications to use to make such
+decisions.
+
+See Håvard’s “Google Transport-Wide Congestion Control” talk for more
+information about this feature.
+
+GstTranscoder: a new high-level transcoding API for applications
+
+The new GstTranscoder library, along with transcodebin and
+uritranscodebin elements, provides high level API for applications to
+transcode media files from one format to another. Watch Thibault’s talk
+“GstTranscoder: A High Level API to Quickly Implement Transcoding
+Capabilities in your Applications” for more information.
+
+This also comes with a gst-transcoder-1.0 command line utility to
+transcode one URI into another URI based on the specified encoding
+profile.
+
+Active Format Description (AFD) and Bar Data support
+
+The GstVideo Ancillary Data API has gained support for Active Format
+Description (AFD) and Bar data.
+
+This includes various two new buffer metas: GstVideoAFDMeta and
+GstVideoBarMeta.
+
+GStreamer now also parses and extracts AFD/Bar data in the h264/h265
+video parsers, and supports both capturing them and outputting them in
+the decklink elements. See Aaron’s lightning talk at the GStreamer
+Conference for more background.
+
+ONVIF trick modes support in both GStreamer RTSP server and client
+
+- Support for the various trick modes described in section 6 of the
+ ONVIF streaming spec has been implemented in both gst-rtsp-server
+ and rtspsrc.
+- Various new properties in rtspsrc must be set to take advantage of
+ the ONVIF support
+- Examples are available here: test-onvif-server.c and
+ test-onvif-client.c
+- Watch Mathieu Duponchelle’s talk “Implementing a Trickmode Player
+ with ONVIF, RTSP and GStreamer” for more information and a live
+ demo.
+
+GStreamer Codecs library with decoder base classes
+
+This introduces a new library in gst-plugins-bad which contains a set of
+base classes that handle bitstream parsing and state tracking for the
+purpose of decoding different codecs. Currently H264, H265, VP8 and VP9
+are supported. These bases classes are meant primarily for internal use
+in GStreamer and are used in various decoder elements in connection with
+low level decoding APIs like DXVA, NVDEC, VAAPI and V4L2 State Less
+decoders. The new library is named gstreamer-codecs-1.0 /
+libgstcodecs-1.0 and is not yet guaranteed to be API stable across major
+versions.
+
+MPEG-TS muxing improvements
+
+The GStreamer MPEG-TS muxer has seen major improvements on various
+fronts in this cycle:
+
+- It has been ported to the GstAggregator base class which means it
+ can work in defined-latency mode with live input sources and
continue streaming if one of the inputs stops producing data.
-- jpegenc has gained a "snapshot" property just like pngenc to make it
- easier to output just a single encoded frame.
-
-- jpegdec will now handle interlaced MJPEG streams properly and also
- handles frames without an End of Image marker better.
-
-- v4l2: There are now video encoders for VP8, VP9, MPEG4, and H263.
- The v4l2 video decoder handles dynamic resolution changes, and the
- video4linux device provider now does much faster device probing. The
- plugin also no longer uses the libv4l2 library by default, as it has
- prevented a lot of interesting use cases like CREATE_BUFS, DMABuf,
- usage of TRY_FMT. As the libv4l2 library is totally inactive and not
- really maintained, we decided to disable it. This might affect a
- small number of cheap/old webcams with custom vendor formats for
- which we do not provide conversion in GStreamer. It is possible to
- re-enable support for libv4l2 at run-time however, by setting the
- environment variable GST_V4L2_USE_LIBV4L2=1.
-
-- rtspsrc now has support for RTSP protocol version 2.0 as well as
- ONVIF audio backchannels (see below for more details). It also
- sports a new "accept-certificate" signal for "manually" checking a
- TLS certificate for validity. It now also prints RTSP/SDP messages
- to the gstreamer debug log instead of stdout.
-
-- shout2send now uses non-blocking I/O and has a configurable network
- operations timeout.
-
-- splitmuxsink has gained a "split-now" action signal and new
- "alignment-threshold" and "use-robust-muxing" properties. If robust
- muxing is enabled, it will check and set the muxer's reserved space
- properties if present. This is primarily for use with mp4mux's
- robust muxing mode.
-
-- qtmux has a new _prefill recording mode_ which sets up a moov header
- with the correct sample positions beforehand, which then allows
- software like Adobe Premiere and FinalCut Pro to import the files
- while they are still being written to. This only works with constant
- framerate I-frame only streams, and for now only support for ProRes
- video and raw audio is implemented. Adding support for additional
- codecs is just a matter of defining appropriate maximum frame sizes
- though.
-
-- qtmux also supports writing of svmi atoms with stereoscopic video
- information now. Trak timescales can be configured on a per-stream
- basis using the "trak-timescale" property on the sink pads. Various
- new formats can be muxed: MPEG layer 1 and 2, AC3 and Opus, as well
- as PNG and VP9.
-
-- souphttpsrc now does connection sharing by default: it shares its
- SoupSession with other elements in the same pipeline via a
- GstContext if possible (session-wide settings are all the defaults).
- This allows for connection reuse, cookie sharing, etc. Applications
- can also force a context to use. In other news, HTTP headers
- received from the server are posted as element messages on the bus
- now for easier diagnostics, and it's also possible now to use other
- types of proxy servers such as SOCKS4 or SOCKS5 proxies, support for
- which is implemented directly in gio. Before only HTTP proxies were
- allowed.
-
-- qtmux, mp4mux and matroskamux will now refuse caps changes of input
- streams at runtime. This isn't really supported with these
- containers (or would have to be implemented differently with a
- considerable effort) and doesn't produce valid and spec-compliant
- files that will play everywhere. So if you can't guarantee that the
- input caps won't change, use a container format that does support on
- the fly caps changes for a stream such as MPEG-TS or use
- splitmuxsink which can start a new file when the caps change. What
- would happen before is that e.g. rtph264depay or rtph265depay would
- simply send new SPS/PPS inband even for AVC format, which would then
- get muxed into the container as if nothing changed. Some decoders
- will handle this just fine, but that's often more luck than by
- design. In any case, it's not right, so we disallow it now.
-
-- matroskamux has Table of Content (TOC) support now (chapters etc.)
- and matroskademux TOC support has been improved. matroskademux has
- also seen seeking improvements searching for the right cluster and
- position.
-
-- videocrop now uses GstVideoCropMeta if downstream supports it, which
- means cropping can be handled more efficiently without any copying.
-
-- compositor now has support for _crossfade blending_, which can be
- used via the new "crossfade-ratio" property on the sink pads.
-
-- The avwait element has a new "end-timecode" property and posts
- "avwait-status" element messages now whenever avwait starts or stops
- passing through data (e.g. because target-timecode and end-timecode
- respectively have been reached).
-
-- h265parse and h265parse will try harder to make upstream output the
- same caps as downstream requires or prefers, thus avoiding
- unnecessary conversion. The parsers also expose chroma format and
- bit depth in the caps now.
-
-- The dtls elements now longer rely on or require the application to
- run a GLib main loop that iterates the default main context
- (GStreamer plugins should never rely on the application running a
- GLib main loop).
-
-- openh264enc allows to change the encoding bitrate dynamically at
- runtime now
-
-- nvdec is a new plugin for hardware-accelerated video decoding using
- the NVIDIA NVDEC API (which replaces the old VDPAU API which is no
- longer supported by NVIDIA)
-
-- The NVIDIA NVENC hardware-accelerated video encoders now support
- dynamic bitrate and preset reconfiguration and support the I420
- 4:2:0 video format. It's also possible to configure the gop size via
- the new "gop-size" property.
-
-- The MPEG-TS muxer and demuxer (tsmux, tsdemux) now have support for
- JPEG2000
-
-- openjpegdec and jpeg2000parse support 2-component images now (gray
- with alpha), and jpeg2000parse has gained limited support for
- conversion between JPEG2000 stream-formats. (JP2, J2C, JPC) and also
- extracts more details such as colorimetry, interlace-mode,
- field-order, multiview-mode and chroma siting.
-
-- The decklink plugin for Blackmagic capture and playback cards have
- seen numerous improvements:
-
-- decklinkaudiosrc and decklinkvideosrc now put hardware reference
- timestamp on buffers in form of GstReferenceTimestampMetas.
- This can be useful to know on multi-channel cards which frames from
- different channels were captured at the same time.
-
-- decklinkvideosink has gained support for Decklink hardware keying
- with two new properties ("keyer-mode" and "keyer-level") to control
- the built-in hardware keyer of Decklink cards.
-
-- decklinkaudiosink has been re-implemented around GstBaseSink instead
- of the GstAudioBaseSink base class, since the Decklink APIs don't
- fit very well with the GstAudioBaseSink APIs, which used to cause
- various problems due to inaccuracies in the clock calculations.
- Problems were audio drop-outs and A/V sync going wrong after
- pausing/seeking.
-
-- support for more than 16 devices, without any artificial limit
-
-- work continued on the msdk plugin for Intel's Media SDK which
- enables hardware-accelerated video encoding and decoding on Intel
- graphics hardware on Windows or Linux. Added the video memory,
- buffer pool, and context/session sharing support which helps to
- improve the performance and resource utilization. Rendernode support
- is in place which helps to avoid the constraint of having a running
- graphics server as DRM-Master. Encoders are exposing a number rate
- control algorithms now. More encoder tuning options like
- trellis-quantiztion (h264), slice size control (h264), B-pyramid
- prediction(h264), MB-level bitrate control, frame partitioning and
- adaptive I/B frame insertion were added, and more pixel formats and
- video codecs are supported now. The encoder now also handles
- force-key-unit events and can insert frame-packing SEIs for
- side-by-side and top-bottom stereoscopic 3D video.
-
-- dashdemux can now do adaptive trick play of certain types of DASH
- streams, meaning it can do fast-forward/fast-rewind of normal (non-I
- frame only) streams even at high speeds without saturating network
- bandwidth or exceeding decoder capabilities. It will keep statistics
- and skip keyframes or fragments as needed. See Sebastian's blog post
- _DASH trick-mode playback in GStreamer_ for more details. It also
- supports webvtt subtitle streams now and has seen improvements when
- seeking in live streams.
-
-- kmssink has seen lots of fixes and improvements in this cycle,
- including:
-
-- Raspberry Pi (vc4) and Xilinx DRM driver support
-
-- new "render-rectangle" property that can be used from the command
- line as well as "display-width" and "display-height", and
- "can-scale" properties
-
-- GstVideoCropMeta support
+- atscmux, a new ATSC-specific tsmux subclass
+
+- Constant Bit Rate (CBR) muxing support via the new bitrate property
+ which allows setting the target bitrate in bps. If this is set the
+ muxer will insert null packets as padding to achieve the desired
+ multiplex-wide constant bitrate.
+
+- compliance fixes for TV broadcasting use cases (esp. ATSC). See
+ Jan’s talk “TV Broadcast compliant MPEG-TS” for details.
+
+- Streams can now be added and removed at runtime: Until now, any
+ streams in tsmux had to be present when the element started
+ outputting its first buffer. Now they can appear at any point during
+ the stream, or even disappear and reappear later using the same PID.
+
+- new pcr-interval property allows applications to configure the
+ desired interval instead of hardcoding it
+
+- basic SCTE-35 support. This is enabled by setting the scte-35-pid
+ property on the muxer. Sending SCTE-35 commands is then done by
+ creating the appropriate SCTE-35 GstMpegtsSection and sending them
+ on the muxer.
+
+- MPEG-2 AAC handling improvements
+
+New elements
+
+- New qmlgloverlay element for rendering a QtQuick scene over the top
+ of a video stream. qmlgloverlay requires that Qt support adopting an
+ external OpenGL context and is known to work on X11 and Windows.
+ Wayland is known not to work due to limitations within Qt. Check out
+ the example to see how it works.
+
+- The clocksync element is a generic element that can be placed in a
+ pipeline to synchronise passing buffers to the clock at that point.
+ This is similar to identity sync=true, but because it isn’t
+ GstBaseTransform-based, it can process GstBufferLists without
+ breaking them into separate GstBuffers. It is also more discoverable
+ than the identity option. Note that you do not need to insert this
+ element into your pipeline to make GStreamer sync to the pipeline
+ clock, this is usually handled automatically by the elements in the
+ pipeline (sources and sinks mostly). This element is useful to feed
+ non-live input such as local files into elements that expect live
+ input such as webrtcbin.`
+
+- New imagesequencesrc element to easily create a video stream from a
+ sequence of JPEG or PNG images (or any other encoding where the type
+ can be detected), basically a multifilesrc made specifically for
+ image sequences.
+
+- rpicamsrc element for capturing raw or encoded video (H.264, MJPEG)
+ from the Raspberry Pi camera. This works much like the popular
+ raspivid command line utility but outputs data nicely timestamped
+ and formatted in order to integrate nicely with other GStreamer
+ elements. Also comes with a device provider so applications can
+ discover the camera if available.
+
+- aatv and cacatv video filters that transform video ASCII art style
+
+- avtp: new Audio Video Transport Protocol (AVTP) plugin for Linux.
+ See Andre Guedes’ talk “Audio/Video Bridging (AVB) support in
+ GStreamer” for more details.
+
+- clockselect: a pipeline element that enables clock selection/forcing
+ via gst-launch pipeline syntax.
+
+- dashsink: Add new sink to produce DASH content. See Stéphane’s talk
+ or blog post for details.
+
+- dvbsubenc: a DVB subtitle encoder element
+
+- microdns: a libmicrodns-based mdns device provider to discover RTSP
+ cameras on the local network
+
+- mlaudiosink: new audio sink element for the Magic Leap platform,
+ accompanied by an MLSDK implementation in the amc plugin
+
+- msdkvp9enc: VP9 encoder element for the Intel MediaSDK
+
+- rist: new plugin implementing support for the Video Services Forum’s
+ Reliable Internet Stream Transport (RIST) TR-06-1 Simple Profile.
+ See Nicolas’ blog post “GStreamer support for the RIST
+ Specification” for more details.
+
+- rtmp2: new RTMP client source and sink elements with fully
+ asynchronous network operations, better robustness and additional
+ features such as handling ping and stats messages, and adobe-style
+ authentication. The new rtmp2src and rtmp2sink elements should be
+ API-compatible with the old rtmpsrc / rtmpsink elements and should
+ work as drop-in replacements.
+
+- new RTP source and sink elements to easily set up RTP streaming via
+ rtp:// URIs: The rtpsink and rtpsrc elements add an URI interface so
+ that streams can be decoded with decodebin using rtp:// URIs. These
+ can be used as follows: ``` gst-launch-1.0 videotestsrc ! x264enc !
+ rtph264pay config-interval=3 ! rtpsink uri=rtp://239.1.1.1:1234
+
+ gst-launch-1.0 videotestsrc ! x264enc ! rtph264pay config-interval=1
+ ! rtpsink uri=rtp://239.1.2.3:5000 gst-launch-1.0 rtpsrc
+ uri=rtp://239.1.2.3:5000?encoding-name=H264 ! rtph264depay !
+ avdec_h264 ! videoconvert ! xvimagesink
+
+ gst-launch-1.0 videotestsrc ! avenc_mpeg4 ! rtpmp4vpay
+ config-interval=1 ! rtpsink uri=rtp://239.1.2.3:5000 gst-launch-1.0
+ rtpsrc uri=rtp://239.1.2.3:5000?encoding-name=MP4V-ES ! rtpmp4vdepay
+ ! avdec_mpeg4 ! videoconvert ! xvimagesink ```
+
+- svthevcenc: new SVT-HEVC-based H.265 video encoder
+
+- switchbin: new helper element which chooses between a set of
+ processing chains (paths) based on input caps, and changes the
+ active chain if new caps arrive. Paths are child objects, which are
+ accessed by the GstChildProxy interface. See the switchbin
+ documentation for a usage example.
+
+- vah264dec: new experimental va plugin with an element for H.264
+ decoding with VA-API using GStreamer’s new stateless decoder
+ infrastructure (see Linux section below).
+
+- v4l2codecs: introduce an V4L2 CODECs Accelerator supporting the new
+ CODECs uAPI in the Linux kernel (see Linux section below)
+
+- zxing new plugin to detect QR codes and barcodes, based on libzxing
+
+- also see the Rust plugins section below which contains plenty of new
+ exciting plugins written in Rust!
+
+New element features and additions
+
+GStreamer core
+
+- filesink: Add a new “full” buffer mode. Previously the default and
+ full modes were the same. Now the default mode is like before: it
+ accumulates all buffers in a buffer list until the threshold is
+ reached and then writes them all out, potentially in multiple
+ writes. The new full mode works by always copying memory to a single
+ memory area and writing everything out with a single write once the
+ threshold is reached.
+
+- multiqueue: Add stats property and
+ current-level-{buffers, bytes, time} pad properties to query the
+ current levels of the corresponding internal queue.
+
+Plugins Base
+
+- alsa: implement a device provider
+
+- alsasrc: added use-driver-timestamp property to force use of
+ pipeline timestamps (and disable driver timestamps) if so desired
+
+- audioconvert: fix changing the mix-matrix property at runtime
+
+- appsrc: added support for segment forwarding or custom GstSegments
+ via GstSample, enabled via the handle-segment-change property. This
+ only works for segments in TIME format for now.
+
+- compositor: various performance optimisations, checkerboard drawing
+ fixes, and support for VUYA format
+
+- encodebin: Fix and refactor smart encoding; ensure that a single
+ segment is pushed into encoders; improve force-key-unit event
+ handling.
+
+- opusenc: Add low delay option (audio-type=restricted-lowdelay) to
+ disable the SILK layer and achieve only 5ms delay.
+
+- opusdec: add stats property to retrieve various decoder statistics.
+
+- uridecodebin3: Let decodebin3 do its stream selection if no one
+ answers
+
+- decodebin3: Avoid overriding explicit user selection of streams
+
+- playbin: add flag to force use of software decoders over any
+ hardware decoders that might also be available
+
+- playbin3, playbin: propagate sink context
+
+- rawvideoparse: Fix tiling support, allow setting colorimetry
+
+- subparse: output plain utf8 text instead of pango-markup formatted
+ text if downstream requires it, useful for interop with elements
+ that only accept utf8-formatted subtitles such as muxers or closed
+ caption converters.
+
+- tcpserversrc, tcpclientsrc: add stats property with TCP connection
+ stats (some are only available on Linux though)
+
+- timeoverlay: add show-times-as-dates, datetime-format and
+ datetime-epoch properties to display times with dates
+
+- videorate: Fix changing rate property during playback; reverse
+ playback fixes; update QoS events taking into account our rate
+
+- videoscale: pass through and transform size sensitive metas instead
+ of just dropping them
+
+Plugins Good
+
+- avidemux can handle H.265 video now. Our advice remains to
+ immediately cease all contact and communication with anyone who
+ hands you H.265 video in an AVI container, however.
+
+- avimux: Add support for S24LE and S32LE raw audio and v210 raw video
+ formats; support more than 2 channels of raw audio.
+
+- souphttpsrc: disable session sharing and cookie jar when the cookies
+ property is set; correctly handle seeks past the end of the content
+
+- deinterlace: new YADIF deinterlace method which should provide
+ better quality than the existing methods and is LGPL licensed;
+ alternate fields are supported as input to the deinterlacer as well
+ now, and there were also fixes for switching the deinterlace mode on
+ the fly.
+
+- flvmux: in streamable mode allow adding new pads even if the initial
+ header has already been written. Old clients will only process the
+ initial stream, new clients will get a header with the new streams.
+ The skip-backwards-streams property can be used to force flvmux to
+ skip and drop a few buffers rather than produce timestamps that go
+ backward and confuse librtmp-based clients. There’s also better
+ handling for timestamp rollover when streaming for a long time.
+
+- imagefreeze: Add live mode, which can be enabled via the new is-live
+ property. In this mode frames will only be output in PLAYING state
+ according to the negotiated framerate, skipping frames if the output
+ can’t keep up (e.g. because it’s blocked downstream). This makes it
+ possible to actually use imagefreeze in live pipelines without
+ having to manually ensure somehow that it starts outputting at the
+ current running time and without still risking to fall behind
+ without recovery.
+
+- matroskademux, qtdemux: Provide audio lead-in for some lossy formats
+ when doing accurate seeks, to make sure we can actually decode
+ samples at the desired position. This is especially important for
+ non-linear audio/video editing use-cases.
+
+- matroskademux, matroskamux: Handle interlaced field order (tff, bff)
+
+- matroskamux:
+
+ - new offset-to-zero property to offset all streams to start at
+ zero. This takes the timestamp of the earliest stream and
+ offsets it so that it starts at 0. Some software (VLC,
+ ffmpeg-based) does not properly handle Matroska files that start
+ at timestamps much bigger than zero, which could happen with
+ live streams.
+ - added a creation-time property to explicitly set the creation
+ time to write into the file headers. Useful when remuxing, for
+ example, but also for live feeds where the DateUTC header can be
+ set a UTC timestamp corresponding to the beginning of the file.
+ - the muxer now also always waits for caps on sparse streams, and
+ warns if caps arrive after the header has already been sent,
+ otherwise the subtitle track might be silently absent in the
+ final file. This might affect applications that send sparse data
+ into matroskamux via an appsrc element, which will usually not
+ send out the initial caps before it sends out the first buffer.
+
+- pulseaudio: device provider improvements: fix discovery of
+ newly-added devices and hide the alsa device provider if we provide
+ alsa devices
+
+- qtdemux: raw audio handling improvements, support for AC4 audio, and
+ key-units trickmode interval support
+
+- qtmux:
+
+ - was ported to the GstAggregator base class which allows for
+ better handling of live inputs, but might entail minor
+ behavioural changes for sparse inputs if inputs are not live.
+ - has also gained a force-create-timecode-trak property to create
+ a timecode trak in non-mov flavors, which may not be supported
+ by Apple but is supported by other software such as Final Cut
+ Pro X
+ - also a force-chunks property to force the creation of chunks
+ even in single-stream files, which is required for Apple ProRes
+ certification.
+ - also supports 8k resolutions in prefill mode with ProRes.
+
+- rtpbin gained a request-jitterbuffer signal which allows
+ applications to plug in their own jitterbuffer implementation such
+ as the threadsharing jitterbuffer from the Rust plugins, for
+ example.
+
+- rtprtxsend: add clock-rate-map property to allow generic RTP input
+ caps without a clock-rate whilst still supporting the max-size-time
+ property for bundled streams.
+
+- rtpssrcdemux: introduce max-streams property to guard against
+ attacks where the sender changes SSRC for every RTP packet.
+
+- rtph264pay, rtph264pay: implement STAP-A and various aggregation
+ modes controled by the new aggegrate-mode property: none to not
+ aggregate NAL units (as before), zero-latency to aggregate NAL units
+ until a VCL or suffix unit is included, or max to aggregate all NAL
+ units with the same timestamp (which adds one frame of latency). The
+ default has been kept at none for backwards compatibility reasons
+ and because various RTP/RTSP implementions don’t handle aggregation
+ well. For WebRTC use cases this should be set to zero-latency,
+ however.
+
+- rtpmp4vpay: add support for config-interval=-1 to resend headers
+ with each IDR keyframe, like other video payloaders.
+
+- rtpvp8depay: Add wait-for-keyframe property for waiting until the
+ next keyframe after packet loss. Useful if the video stream was not
+ encoded with error resilience enabled, in which case packet loss
+ tends to cause very bad artefacts when decoding, and waiting for the
+ next keyframe instead improves user experience considerably.
+
+- splitmuxsink and splitmuxsrc can now handle auxiliary video streams
+ in addition to the primary video stream. The primary video stream is
+ still used to select fragment cut points at keyframe boundaries.
+ Auxilliary video streams may be broken up at any packet - so
+ fragments may not start with a keyframe for those streams.
+
+- splitmuxsink:
+
+ - new muxer-preset and sink-preset properties for setting
+ muxer/sink presets
+ - a new start-index property to set the initial fragment id
+ - and a new muxer-pad-map property which explicitly maps
+ splitmuxsink pads to the muxer pads they should connect to,
+ overriding the implicit logic that tries to match pads but
+ yields arbitrary names.
+ - Also includes the actual sink element in the fragment-opened and
+ fragment-closed element messages now, which is especially useful
+ for sinks without a location property or when finalisation of
+ the fragments is done asynchronously.
+
+- videocrop: add support for Y444, Y41B and Y42B pixel formats
+
+- vp8enc, vp9enc: change default value of VP8E_SET_STATIC_THRESHOLD
+ from 0 to 1 which matches what Google WebRTC does and results in
+ lower CPU usage; also added a new bit-per-pixel property to select a
+ better default bitrate
+
+- v4l2: add support for ABGR, xBGR, RGBA, and RGBx formats and for
+ handling interlaced video in alternate fields interlace mode (one
+ field per buffer instead of one frame per picture with both fields
+ interleaved)
+
+- v4l2: Profile and level probing support for H264, H265, MPEG-4,
+ MPEG-2, VP8, and VP9 video encoders and decoders
+
+Plugins Ugly
+
+- asfdemux: extract more metadata: disc number and disc count
+
+- x264enc:
+
+ - respect YouTube bitrate recommendation when user sets the
+ YouTube profile preset
+ - separate high-10 video formats from 8-bit formats to improve
+ depth negotiation and only advertise suitable input raw formats
+ for the desired output depth
+ - forward downstream colorimetry and chroma-site restrictions to
+ upstream elements
+ - support more color primaries/mappings
+
+Plugins Bad
+
+- av1enc: add threads, row-mt and tile-{columns,rows} properties for
+ this AOMedia AV1 encoder
+
+- ccconverter: implement support for CDP framerate conversions
+
+- ccextractor: Add remove-caption-meta property to remove caption
+ metas from the outgoing video buffers
+
+- decklink: add support for 2K DCI video modes, widescreen NTSC/PAL,
+ and for parsing/outputting AFD/Bar data. Also implement a simple
+ device provider for Decklink devices.
+
+- dtlsrtpenc: add rtp-sync property which synchronises RTP streams to
+ the pipeline clock before passing them to funnel for merging with
+ RTCP.
+
+- fdkaac: also decode MPEG-2 AAC; encoder now supports more
+ multichannel/surround sound layouts
+
+- hlssink2: add action signals for custom playlist/fragment handling:
+ Instead of always going through the file system API we allow the
+ application to modify the behaviour. For the playlist itself and
+ fragments, the application can provide a GOutputStream. In addition
+ the sink notifies the application whenever a fragment can be
+ deleted.
+
+- interlace: can now output data in alternate fields mode; added field
+ switching mode for 2:2 field pattern
+
+- iqa: Add a mode property to enable strict mode that checks that all
+ the input streams have the exact same number of frames; also
+ implement the child proxy interface
+
+- mpeg2enc: add disable-encode-retries property for lower CPU usage
+
+- mpeg4videoparse: allow re-sending codec config at IDR via
+ config-interval=-1
+
+- mpegtsparse: new alignment property to determine number of TS
+ packets per output buffer, useful for feeding an MPEG-TS stream for
+ sending via udpsink. This can be used in combination with the
+ split-on-rai property that makes sure to start a new output buffer
+ for any TS packet with the Random Access Indicator set. Also set
+ delta unit buffer flag on non-random-access buffers.
+
+- mpegdemux: add an ignore-scr property to ignore the SCR in
+ non-compliant MPEG-PS streams with a broken SCR, which will work as
+ long as PTS/DTS in the PES header is consistently increasing.
+
+- tsdemux:
+
+ - add an ignore-pcr property to ignore MPEG-TS streams with broken
+ PCR streams on which we can’t reliably recover correct
+ timestamps.
+ - new latency property to allow applications to lower the
+ advertised worst-case latency of 700ms if they know their
+ streams support this (must have timestamps in higher frequency
+ than required by the spec)
+ - support for AC4 audio
+
+- msdk - Intel Media SDK plugin for hardware-accelerated video
+ decoding and encoding on Windows and Linux:
+
+ - mappings for more video formats: Y210, Y410, P012_LE, Y212_LE
+ - encoders now support bitrate changes and input format changes in
+ playing state
+ - msdkh264enc, msdkh265enc: add support for CEA708 closed caption
+ insertion
+ - msdkh264enc, msdkh265enc: set Region of Interest (ROI) region
+ from ROI metas
+ - msdkh264enc, msdkh265enc: new tune property to enable low-power
+ mode
+ - msdkh265enc: add support 12-bit 4:2:0 encoding and 8-bit 4:2:2
+ encoding and VUYA, Y210, and Y410 as input formats
+ - msdkh265enc: add support for screen content coding extension
+ - msdkh265dec: add support for main-12/main-12-intra,
+ main-422-10/main-422-10-intra 10bit,
+ main-422-10/main-422-10-intra 8bit,
+ main-422-12/main-422-12-intra, main-444-10/main-444-10-intra,
+ main-444-12/main-444-12-intra, and main-444 profiles
+ - msdkvp9dec: add support for 12-bit 4:4:4
+ - msdkvpp: add support for Y410 and Y210 formats, cropping via
+ properties, and a new video-direction property.
+
+- mxf: Add support for CEA-708 CDP from S436 essence tracks. mxfdemux
+ can now handle Apple ProRes
+
+- nvdec: add H264 + H265 stateless codec implementation nvh264sldec
+ and nvh265sldec with fewer features but improved latency. You can
+ set the environment variable GST_USE_NV_STATELESS_CODEC=h264 to use
+ the stateless decoder variant as nvh264dec instead of the “normal”
+ NVDEC decoder implementation.
+
+- nvdec: add support for 12-bit 4:4:4/4:2:0 and 10-bit 4:2:0 decoding
+
+- nvenc:
+
+ - add more rate-control options, support for B-frame encoding (if
+ device supports it), an aud property to toggle Access Unit
+ Delimiter insertion, and qp-{min,max,const}-{i,p,b} properties.
+ - the weighted-pred property enables weighted prediction.
+ - support for more input formats, namely 8-bit and 10-bit RGB
+ formats (BGRA, RGBA, RGB10A2, BGR10A2) and YV12 and VUYA.
+ - on-the-fly resolution changes are now supported as well.
+ - in case there are multiple GPUs on the system, there are also
+ per-GPU elements registered now, since different devices will
+ have different capabilities.
+ - nvh265enc can now support 10-bit YUV 4:4:4 encoding and 8-bit
+ 4:4:4 / 10-bit 4:2:0 formats up to 8K resolution (with some
+ devices). In case of HDR content HDR related SEI nals will be
+ inserted automatically.
+
+- openjpeg: enable multi-threaded decoding and add support for
+ sub-frame encoding (for lower latency)
+
+- rtponviftimestamp: add opt-out “drop-out-of-segment” property
+
+- spanplc: new stats property
+
+- srt: add support for IPv6 and for using hostnames instead of IP
+ addresses; add streamid property, but also allow passing the id via
+ the stream URI; add wait-for-connection property to srtsink
+
+- timecodestamper: this element was rewritten with an updated API
+ (properties); it has gained many new properties, seeking support and
+ support for linear timecode (LTC) from an audio stream.
+
+- uvch264src now comes with a device provider to advertise available
+ camera sources that support this interface (mostly Logitech C920s)
+
+- wpe: Add software rendering support and support for mouse scroll
+ events
+
+- x265enc: support more 8/10/12 bits 4:2:0, 4:2:2 and 4:4:4 profiles;
+ add support for mastering display info and content light level
+ encoding SEIs
+
+gst-libav
+
+- Add mapping for SpeedHQ video codec used by NDI
+
+- Add mapping for aptX and aptX-HD
+
+- avivf_mux: support VP9 and AV1
+
+- avvidenc: shift output buffer timestamps and output segment by 1h
+ just like x264enc does, to allow for negative DTS.
+
+- avviddec: Limit default number of decoder threads on systems with
+ more than 16 cores, as the number of threads used in avdec has a
+ direct impact on the latency of the decoder, which is of as many
+ frames as threads, so a large numbers of threads can make for
+ latency levels that can be problematic in some applications.
+
+- avviddec: Add thread-type property that allows applications to
+ specify the preferred multithreading method (auto, frame, slice).
+ Note that thread-type=frame may introduce additional latency
+ especially in live pipelines, since it introduces a decoding delay
+ of number of thread frames.
Plugin and library moves
-MPEG-1 audio (mp1, mp2, mp3) decoders and encoders moved to -good
-
-Following the expiration of the last remaining mp3 patents in most
-jurisdictions, and the termination of the mp3 licensing program, as well
-as the decision by certain distros to officially start shipping full mp3
-decoding and encoding support, these plugins should now no longer be
-problematic for most distributors and have therefore been moved from
--ugly and -bad to gst-plugins-good. Distributors can still disable these
-plugins if desired.
-
-In particular these are:
-
-- mpg123audiodec: an mp1/mp2/mp3 audio decoder using libmpg123
-- lamemp3enc: an mp3 encoder using LAME
-- twolamemp2enc: an mp2 encoder using TwoLAME
-
-GstAggregator moved from -bad to core
-
-GstAggregator has been moved from gst-plugins-bad to the base library in
-GStreamer and is now stable API.
-
-GstAggregator is a new base class for mixers and muxers that have to
-handle multiple input pads and aggregate streams into one output stream.
-It improves upon the existing GstCollectPads API in that it is a proper
-base class which was also designed with live streaming in mind.
-GstAggregator subclasses will operate in a mode with defined latency if
-any of the inputs are live streams. This ensures that the pipeline won't
-stall if any of the inputs stop producing data, and that the configured
-maximum latency is never exceeded.
-
-GstAudioAggregator, audiomixer and audiointerleave moved from -bad to -base
-
-GstAudioAggregator is a new base class for raw audio mixers and muxers
-and is based on GstAggregator (see above). It provides defined-latency
-mixing of raw audio inputs and ensures that the pipeline won't stall
-even if one of the input streams stops producing data.
-
-As part of the move to stabilise the API there were some last-minute API
-changes and clean-ups, but those should mostly affect internal elements.
-
-It is used by the audiomixer element, which is a replacement for
-'adder', which did not handle live inputs very well and did not align
-input streams according to running time. audiomixer should behave much
-better in that respect and generally behave as one would expected in
-most scenarios.
-
-Similarly, audiointerleave replaces the 'interleave' element which did
-not handle live inputs or non-aligned inputs very robustly.
-
-GstAudioAggregator and its subclases have gained support for input
-format conversion, which does not include sample rate conversion though
-as that would add additional latency. Furthermore, GAP events are now
-handled correctly.
-
-We hope to move the video equivalents (GstVideoAggregator and
-compositor) to -base in the next cycle, i.e. for 1.16.
-
-GStreamer OpenGL integration library and plugin moved from -bad to -base
-
-The GStreamer OpenGL integration library and opengl plugin have moved
-from gst-plugins-bad to -base and are now part of the stable API canon.
-Not all OpenGL elements have been moved; a few had to be left behind in
-gst-plugins-bad in the new openglmixers plugin, because they depend on
-the GstVideoAggregator base class which we were not able to move in this
-cycle. We hope to reunite these elements with the rest of their family
-for 1.16 though.
-
-This is quite a milestone, thanks to everyone who worked to make this
-happen!
-
-Qt QML and GTK plugins moved from -bad to -good
-
-The Qt QML-based qmlgl plugin has moved to -good and provides a
-qmlglsink video sink element as well as a qmlglsrc element. qmlglsink
-renders video into a QQuickItem, and qmlglsrc captures a window from a
-QML view and feeds it as video into a pipeline for further processing.
-Both elements leverage GStreamer's OpenGL integration. In addition to
-the move to -good the following features were added:
-
-- A proxy object is now used for thread-safe access to the QML widget
- which prevents crashes in corner case scenarios: QML can destroy the
- video widget at any time, so without this we might be left with a
- dangling pointer.
-
-- EGL is now supported with the X11 backend, which works e.g. on
- Freescale imx6
-
-The GTK+ plugin has also moved from -bad to -good. It includes gtksink
-and gtkglsink which both render video into a GtkWidget. gtksink uses
-Cairo for rendering the video, which will work everywhere in all
-scenarios but involves an extra memory copy, whereas gtkglsink fully
-leverages GStreamer's OpenGL integration, but might not work properly in
-all scenarios, e.g. where the OpenGL driver does not properly support
-multiple sharing contexts in different threads; on Linux Nouveau is
-known to be broken in this respect, whilst NVIDIA's proprietary drivers
-and most other drivers generally work fine, and the experience with
-Intel's driver seems to be mixed; some proprietary embedded Linux
-drivers don't work; macOS works).
-
-GstPhysMemoryAllocator interface moved from -bad to -base
-
-GstPhysMemoryAllocator is a marker interface for allocators with
-physical address backed memory.
+- There were no plugin moves or library moves in this cycle.
+
+- The rpicamsrc element was moved into -good from an external
+ repository on github.
Plugin removals
-- the sunaudio plugin was removed, since it couldn't ever have been
- built or used with GStreamer 1.0, but no one even noticed in all
- these years.
+The following elements or plugins have been removed:
+
+- The yadif video deinterlacing plugin from gst-plugins-bad, which was
+ one of the few GPL licensed plugins, has been removed in favour of
+ deinterlace method=yadif.
+
+- The avdec_cdgraphics CD Graphics video decoder element from
+ gst-libav was never usable in GStreamer and we now have a cdgdec
+ element written in Rust in gst-plugins-rs to replace it.
+
+- The VDPAU plugin has been unmaintained and unsupported for a very
+ long time and does not have the feature set we expect from
+ hardware-accelerated video decoders. It’s been superseded by the
+ nvcodec plugin leveraging NVIDIA’s NVDEC API.
+
+Miscellaneous API additions
+
+GStreamer core
+
+- gst_task_resume(): This new API allows resuming a task if it was
+ paused, while leaving it in stopped state if it was stopped or not
+ started yet. This can be useful for callback-based driver workflows,
+ where you basically want to pause and resume the task when buffers
+ are notified while avoiding the race with a gst_task_stop() coming
+ from another thread.
+
+- info: add printf extensions GST_TIMEP_FORMAT and GST_STIMEP_FORMAT
+ for printing GstClockTime/GstClockTimeDiff pointers, which is much
+ more convenient to use in debug log statements than the usual
+ GST_TIME_FORMAT-followed-by-GST_TIME_ARGS dance. Also add an
+ explicit GST_STACK_TRACE_SHOW_NONE enum value.
+
+- gst_element_get_current_clock_time() and
+ gst_element_get_current_running_time(): new helper functions for
+ getting an element clock’s time, and the clock time minus base time,
+ respectively. Useful when adding additional input branches to
+ elements such as compositor, audiomixer, flvmux, interleave or
+ input-selector to determine initial pad offsets and such.
+
+- seeking: Add GST_SEEK_FLAG_TRICKMODE_FORWARD_PREDICTED to just skip
+ B-frames during trick mode, showing both keyframes + P-frame, and
+ add support for it in h264parse and h265parse.
+
+- elementfactory: add GST_ELEMENT_FACTORY_TYPE_HARDWARE to allow
+ elements to advertise that they are hardware-based or interact with
+ hardware. This has multiple applications:
+
+ - it makes it possible to easily differentiate hardware and
+ software based element implementations such as audio or video
+ encoders and decoders. This is useful in order to force the use
+ of software decoders for specific use cases, or to check if a
+ selected decoder is actually hardware-accelerated or not.
+ - elements interacting with hardware and their respective drivers
+ typically don’t know the actually supported capabilities until
+ the element is set into at least READY state and can open a
+ device handle and probe the hardware.
+
+- gst_uri_from_string_escaped(): identical to gst_uri_from_string()
+ except that the userinfo and fragment components of the URI will not
+ be unescaped while parsing. This is needed for correctly parsing
+ usernames or passwords with : in them .
+
+- paramspecs: new GstParamSpec flag GST_PARAM_CONDITIONALLY_AVAILABLE
+ to indicate that a property might not always exist.
+
+- gst_bin_iterate_all_by_element_factory_name() finds elements in a
+ bin by factory name
+
+- pad: gst_pad_get_single_internal_link() is a new convenience
+ function to return the single internal link of a pad, which is
+ useful e.g. to retrieve the output pad of a new multiqueue request
+ pad.
+
+- datetime: Add constructors to create datetimes with timestamps in
+ microseconds, gst_date_time_new_from_unix_epoch_local_time_usecs()
+ and gst_date_time_new_from_unix_epoch_utc_usecs().
+
+- gst_debug_log_get_lines() gets debug log lines formatted in the same
+ way the default log handler would print them
+
+- GstSystemClock: Add GST_CLOCK_TYPE_TAI as GStreamer abstraction for
+ CLOCK_TAI, to support transmission offloading features where network
+ packets are timestamped with the time they are deemed to be actually
+ transmitted. Useful in combination with the new AVTP plugin.
+
+- miscellaneous utility functions: gst_clear_uri(),
+ gst_structure_take().
+
+- harness: Added gst_harness_pull_until_eos()
+
+- GstBaseSrc:
+
+ - gst_base_src_new_segment() allows subclasses to update the
+ segment to be used at runtime from the ::create() function. This
+ deprecates gst_base_src_new_seamless_segment()
+ - gst_base_src_negotiate() allows subclasses to trigger format
+ renegotiation at runtime from inside the ::create() or ::alloc()
+ function
+
+- GstBaseSink: new stats property and gst_base_sink_get_stats() method
+ to retrieve various statistics such as average frame rate and
+ dropped/rendered buffers.
+
+- GstBaseTransform: gst_base_transform_reconfigure() is now public
+ API, useful for subclasses that need to completely re-implement the
+ ::submit_input_buffer() virtual method
+
+- GstAggregator:
+
+ - gst_aggregator_update_segment() allows subclasses to update the
+ output segment at runtime. Subclasses should use this function
+ rather than push a segment event onto the source pad directly.
+ - new sample selection API:
+ - subclasses should now call gst_aggregator_selected_samples()
+ from their ::aggregate() implementation to signal that they
+ have selected the next samples they will aggregate
+ - GstAggregator will then emit the samples-selected signal
+ where handlers can then look up samples per pad via
+ gst_aggregator_peek_next_sample().
+ - This is useful for example to atomically update input pad
+ properties in mixer subclasses such as compositor.
+ Applications can now update properties with precise control
+ of when these changes will take effect, and for which input
+ buffer(s).
+ - gst_aggregator_finish_buffer_list() allows subclasses to push
+ out a buffer list, improving efficiency in some cases.
+ - a ::negotiate() virtual method was added, for consistency with
+ other base classes and to allow subclasses to completely
+ override the negotiation behaviour.
+ - the new ::sink_event_pre_queue() and ::sink_query_pre_queue()
+ virtual methods allow subclasses to intercept or handle
+ serialized events and queries before they’re queued up
+ internally.
+
+GStreamer Plugins Base Libraries
+
+Audio library
+
+- audioaggregator, audiomixer: new output-buffer-duration-fraction
+ property which allows use cases such as keeping the buffers output
+ by compositor on one branch and audiomixer on another perfectly
+ aligned, by requiring the compositor to output a n/d frame rate, and
+ setting output-buffer-duration-fraction to d/n on the audiomixer.
+
+- GstAudioDecoder: new max-errors property so applications can
+ configure at what point the decoder should error out, or tell it to
+ just keep going
+
+- gst_audio_make_raw_caps() and gst_audio_formats_raw() are
+ bindings-friendly versions of the GST_AUDIO_CAPS_MAKE() C macro.
+
+- gst_audio_info_from_caps() now handles encoded audio formats as well
+
+PbUtils library
+
+- GstEncodingProfile:
+ - Do not restrict number of similar profiles in a container
+ - add GstValue serialization function
+- codec utils now support more H.264/H.265 profiles/levels and have
+ improved extension handling
+
+RTP library
+
+- rtpbasepayloader: Add scale-rtptime property for scaling RTP
+ timestamp according to the segment rate (equivalent to RTSP speed
+ parameter). This is useful for ONVIF trickmodes via RTSP.
+
+- rtpbasepayload: add experimental property for embedding twcc
+ sequencenumbers for Transport-Wide Congestion Control (gated behind
+ the GST_RTP_ENABLE_EXPERIMENTAL_TWCC_PROPERTY environment
+ variable) - more generic API for enabling this is expected to land
+ in the next development cycle.
+
+- rtcpbuffer: add RTPFB_TYPE_TWCC for Transport-Wide Congestion
+ Control
+
+- rtpbuffer: add
+ gst_rtp_buffer_get_extension_onebyte_header_from_bytes()``, so that one can parse theGBytes`
+ returned by gst_rtp_buffer_get_extension_bytes()
+
+- rtpbasedepayload: Add max-reorder property to make the
+ previously-hardcoded value when to consider a sender to have
+ restarted configurable. In some scenarios it’s particularly useful
+ to set max-reorder=0 to disable the behaviour that the depayloader
+ will drop packets: when max-reorder is set to 0 all
+ reordered/duplicate packets are considered coming from a restarted
+ sender.
+
+RTSP library
+
+- add gst_rtsp_url_get_request_uri_with_control() to create request
+ uri combined with control url
+
+- GstRTSPConnection: add the possibility to limit the Content-Length
+ for RTSP messages via
+ gst_rtsp_connection_set_content_length_limit(). The same
+ functionality is also exposed in gst-rtsp-server.
+
+SDP library
+
+- add support for parsing the extmap attribute from caps and storing
+ inside caps The extmap attribute allows mapping RTP extension header
+ IDs to well-known RTP extension header specifications. See RFC8285
+ for details.
+
+Tags library
+
+- update to latest iso-code and support more languages
+
+- add tags for acoustid id & acoustid fingerprint, plus MusicBrainz ID
+ handling fixes
+
+Video library
+
+- High Dynamic Range (HDR) video information representation and
+ signalling enhancements:
+
+ - New APIs for HDR video information representation and
+ signalling:
+ - GstVideoMasteringDisplayInfo: display color volume info as
+ per SMPTE ST 2086
+ - GstVideoContentLightLevel: content light level specified in
+ CEA-861.3, Appendix A.
+ - plus functions to serialise/deserialise and add them to or
+ parse them from caps
+ - gst_video_color_{matrix,primaries,transfer}_{to,from}_iso():
+ new utilility functions for conversion from/to ISO/IEC
+ 23001-8
+ - add ARIB STD-B67 transfer chracteristic function
+ - add SMPTE ST 2084 support and BT 2100 colorimetry
+ - define bt2020-10 transfer characteristics for clarity:
+ bt707, bt2020-10, and bt2020-12 transfer characteristics are
+ functionally identical but have their own unique values in
+ the specification.
+ - h264parse, h265parse: Parse mastering display info and content
+ light level from SEIs.
+ - matroskademux: parse HDR metadata
+ - matroskamux: Write MasteringMetadata and Max{CLL,FALL}. Enable
+ muxing with HDR meta data if upstream provided it
+ - avviddec: Extract HDR information if any and map bt2020-10, PQ
+ and HLG transfer functions
+
+- added bt601 transfer function (for completeness)
+
+- support for more pixel formats:
+
+ - Y412 (packed 12 bits 4:4:4:4)
+ - Y212 (packed 12 bits 4:2:2)
+ - P012 (semi-planar 4:2:0)
+ - P016_{LE,BE} (semi-planar 16 bits 4:2:0)
+ - Y444_16{LE,BE} (planar 16 bits 4:4:4)
+ - RGB10A2_LE (packed 10-bit RGB with 2-bit alpha channel)
+ - NV12_32L32 (NV12 with 32x32 tiles in linear order)
+ - NV12_4L4 (NV12 with 4x4 tiles in linear order)
+
+- GstVideoDecoder:
+
+ - new max-errors property so applications can configure at what
+ point the decoder should error out, or tell it to just keep
+ going
+
+ - new qos property to disable dropping frames because of QoS, and
+ post QoS messages on the bus when dropping frames. This is
+ useful for example in a scenario where the decoded video is
+ tee-ed off to go into a live sink that syncs to the clock in one
+ branch, and an encoding and save to file pipeline in the other
+ branch. In that case one wouldn’t want QoS events from the video
+ sink make the decoder drop frames because that would also leave
+ gaps in the encoding branch then.
+
+- GstVideoEncoder:
+
+ - gst_video_encoder_finish_subframe() is new API to push out
+ subframes (e.g. slices), so encoders can split the encoding into
+ subframes, which can be useful to reduce the overall end-to-end
+ latency as we no longer need to wait for the full frame to be
+ encoded to start decoding or sending out the data.
+ - new min-force-key-unit-interval property allows configuring the
+ minimum interval between force-key-unit requests and prevents a
+ big bitrate increase if a lot of key-units are requested in a
+ short period of time (as might happen in live streaming RTP
+ pipelines when packet loss is detected).
+ - various force-key-unit event handling fixes
+
+- GstVideoAggregator, compositor, glvideomixer: expose
+ max-last-buffer-repeat property on pads. This can be used to have a
+ compositor display either the background or a stream on a lower
+ zorder after a live input stream freezes for a certain amount of
+ time, for example because of network issues.
+
+- gst_video_format_info_component() is new API to find out which
+ components are packed into a given plane, which is useful to prevent
+ us from assuming a 1-1 mapping between planes and components.
+
+- gst_video_make_raw_caps() and gst_video_formats_raw() are
+ bindings-friendly versions of the GST_VIDEO_CAPS_MAKE() C macro.
+
+- video-blend: Add support for blending on top of 16 bit per component
+ formats, which makes sure we can support every currently supported
+ raw video format for blending subtitles or logos on top of video.
+
+- GST_VIDEO_BUFFER_IS_TOP_FIELD() and
+ GST_VIDEO_BUFFER_IS_BOTTOM_FIELD() convenience macros to check
+ whether the video buffer contains only the top field or bottom field
+ of an interlaced picture.
+
+- GstVideoMeta now includes an alignment field with the
+ GstVideoAlignment so buffer producers can explicitly specify the
+ exact geometry of the planes, allowing users to easily know the
+ padded size and height of each plane. Default values will be used if
+ this is not set.
+
+ Use gst_video_meta_set_alignment() to set the alignment and
+ gst_video_meta_get_plane_size() or gst_video_meta_get_plane_height()
+ to compute the plane sizes or plane heights based on the information
+ in the video meta.
+
+- gst_video_info_align_full() works like gst_video_info_align() but
+ also retrieves the plane sizes.
+
+MPEG-TS library
+
+- support for SCTE-35 sections
+
+- extend support for ATSC tables:
+
+ - System Time Table (STT)
+ - Master Guide Table (MGT)
+ - Rating Region Table (RRT)
+
+Miscellaneous performance, latency and memory optimisations
+
+As always there have been many performance and memory usage improvements
+across all components and modules. Some of them have already been
+mentioned elsewhere so won’t be repeated here.
+
+The following list is only a small snapshot of some of the more
+interesting optimisations that haven’t been mentioned in other contexts
+yet:
+
+- caps negotiation, structure and GValue performance optimizations
+
+- systemclock: clock waiting performance improvements (moved from
+ GstPoll to GCond for waiting), especially on Windows.
+
+- rtpsession: add support for buffer lists on the recv path for better
+ performance with higher packet rate streams.
+
+- rtpjitterbuffer: internal timer handling has been rewritten for
+ better performance, see Nicolas’ talk “Revisiting RTP Jitter Buffer
+ Timers” for more details.
+
+- H.264/H.265 parsers and RTP payloaders/depayloaders have been
+ optimised for latency to make sure data is processed and pushed out
+ as quickly as possible
+
+- video-scaler: correctness and performance improvements, esp. for
+ interlaced formats and GBRA
+
+- GstVideoEncoder has gained new API to push out subframes
+ (e.g. slices), so encoders can split the encoding into subframes,
+ which can be useful to reduce the overall end-to-end latency as we
+ no longer need to wait for the full frame to be encoded to start
+ decoding or sending out the data.
+
+ This is complemented by the new GST_VIDEO_BUFFER_FLAG_MARKER which
+ is a video-specific buffer flag to mark the end of a video frame, so
+ elements can know that they have received all data for a frame
+ without waiting for the beginning of the next frame. This is similar
+ to how the RTP marker flag is used in many RTP video mappings.
+
+ The video encoder base class now also releases the internal stream
+ lock before pushing out data, so as to not block the input side of
+ things from processing more data in the meantime.
+
+Miscellaneous other changes and enhancements
+
+- it is now possible to modify the initial rank of plugin features
+ without modifying the source code or writing code to do so
+ programmatically via the GST_PLUGIN_FEATURE_RANK environment
+ variable. Users can adjust the rank of plugin(s) by passing a
+ comma-separated list of feature:rank pairs where rank can be a
+ numerical value or one of NONE, MARGINAL, SECONDARY, PRIMARY, and
+ MAX. Example: GST_PLUGIN_FEATURE_RANK=myh264dec:MAX,avdec_h264:NONE
+ sets the rank of the myh264dec element feature to the maximum and
+ that of avdec_h264 to 0 (none), thus ensuring that myh264dec is
+ prefered as H264 decoder in an autoplugging context.
-- the schroedinger-based Dirac encoder/decoder plugin has been
- removed, as there is no longer any upstream or anyone else
- maintaining it. Seeing that it's quite a fringe codec it seemed best
- to simply remove it.
+- GstDeviceProvider now does a static probe on start as fallback for
+ providers that don’t support dynamic probing to make things easier
+ for users
-API removals
+WebRTC
-- some MPEG video parser API in the API unstable codecutils library in
- gst-plugins-bad was removed after having been deprecated for 5
- years.
+- webrtcbin now contains initial support for renegotiation involving
+ stream addition and removal. There are a number of caveats to this
+ initial renegotiation support and many complex scenarios are known
+ to require some work.
+- webrtcbin now exposes the internal ICE object for advanced
+ configuration options. Using the internal ICE object, it is possible
+ to toggle UDP or TCP connection usage as well as provide local
+ network addresses.
-Miscellaneous changes
+- Fix a number of call flows within webrtcbin’s GstPromise handling
+ where a promise was never replied to. This has been fixed and now a
+ promise will always receive a reply.
-- The video support library has gained support for a few new pixel
- formats:
-- NV16_10LE32: 10-bit variant of NV16, packed into 32bit words (plus 2
- bits padding)
-- NV12_10LE32: 10-bit variant of NV12, packed into 32bit words (plus 2
- bits padding)
-- GRAY10_LE32: 10-bit grayscale, packed in 32bit words (plus 2 bits
- padding)
+- webrtcbin now exposes a latency property for configuring the
+ internal rtpjitterbuffer latency and buffering when receiving
+ streams.
-- decodebin, playbin and GstDiscoverer have seen stability
- improvements in corner cases such as shutdown while still starting
- up or shutdown in error cases (hat tip to the oss-fuzz project).
+- webrtcbin now only synchronises the RTP part of a stream, allowing
+ RTCP messages to skip synchronisation entirely.
-- floating reference handling was inconsistent and has been cleaned up
- across the board, including annotations. This solves various
- long-standing memory leaks in language bindings, which e.g. often
- caused elements and pads to be leaked.
+- Fixed most of the webrtcbin state properties (connection-state,
+ ice-connection-state, signaling-state, but not ice-gathering-state
+ as that requires newer API in libnice and will be fixed in the next
+ release series) to advance through the state values correctly. Also
+ implemented DTLS connection states in the DTLS elements so that
+ peer-connection-state is not always new.
-- major gobject-introspection annotation improvements for large parts
- of the library API, including nullability of return types and
- function parameters, correct types (e.g. strings vs. filenames),
- ownership transfer, array length parameters, etc. This allows to use
- bigger parts of the GStreamer API to be safely used from dynamic
- language bindings (e.g. Python, Javascript) and allows static
- bindings (e.g. C#, Rust, Vala) to autogenerate more API bindings
- without manual intervention.
+- webrtcbin now accounts for the a=ice-lite attribute in a remote SDP
+ offer and will configure the internal ICE implementation
+ accordingly.
+
+- webrtcbin will now resolve .local candidate addresses using the
+ system DNS resolver. .local candidate addresses are now produced by
+ web browsers to help protect the privacy of users.
+
+- webrtcbin will now add candidates found in the SDP to the internal
+ ICE agent. This was previously unsupported and required using the
+ add-ice-candidate signal manually from the application.
+
+- webrtcbin will now correctly parse a TURN URI that contains a
+ username or password with a : in it.
+
+- The GStreamer WebRTC library gained a GstWebRTCDataChannel object
+ roughly matching the interface exposed by the WebRTC specification
+ to allow for easier binding generation and use of data channels.
OpenGL integration
-- The GStreamer OpenGL integration library has moved to
- gst-plugins-base and is now part of our stable API.
+GStreamer OpenGL bindings/build related changes
-- new MESA3D GBM BACKEND. On devices with working libdrm support, it
- is possible to use Mesa3D's GBM library to set up an EGL context
- directly on top of KMS. This makes it possible to use the GStreamer
- OpenGL elements without a windowing system if a libdrm- and
- Mesa3D-supported GPU is present.
+- The GStreamer OpenGL library (libgstgl) now ships pkg-config files
+ for platform-specific API where libgstgl provides a public
+ integration interface and a pkg-config file for a dependency on the
+ detected OpenGL headers. The new list of pkg-config files in
+ addition to the original gstreamer-gl-1.0 are gstreamer-gl-x11-1.0,
+ gstreamer-gl-wayland-1.0, gstreamer-gl-egl-1.0, and
+ gstreamer-gl-prototypes-1.0 (for OpenGL headers when including
+ gst/gl/gstglfuncs.h).
-- Prefer wayland display over X11: As most Wayland compositors support
- XWayland, the X11 backend would get selected.
+- GStreamer OpenGL now ships some platform-specific introspection data
+ for platforms that have a public interface. This should allow for
+ easier integration with bindings involving platform specific
+ functionality. The new introspection data files are named
+ GstGLX11-1.0, GstGLWayland-1.0, and GstGLEGL-1.0.
-- gldownload can export dmabufs now, and glupload will advertise
- dmabuf as caps feature.
+GStreamer OpenGL Features
+- The iOS implementation no longer accesses UIKit objects off the main
+ thread fixing a loud warning message when used in iOS applications.
-Tracing framework and debugging improvements
+- Support for mouse and keyboard handling using the GstNavigation
+ interface was added for the wayland implementation complementing the
+ already existing support for the X11 and Windows implementations.
-- NEW MEMORY RINGBUFFER BASED DEBUG LOGGER, useful for long-running
- applications or to retrieve diagnostics when encountering an error.
- The GStreamer debug logging system provides in-depth debug logging
- about what is going on inside a pipeline. When enabled, debug logs
- are usually written into a file, printed to the terminal, or handed
- off to a log handler installed by the application. However, at
- higher debug levels the volume of debug output quickly becomes
- unmanageable, which poses a problem in disk-space or bandwidth
- restricted environments or with long-running pipelines where a
- problem might only manifest itself after multiple days. In those
- situations, developers are usually only interested in the most
- recent debug log output. The new in-memory ringbuffer logger makes
- this easy: just installed it with gst_debug_add_ring_buffer_logger()
- and retrieve logs with gst_debug_ring_buffer_logger_get_logs() when
- needed. It is possible to limit the memory usage per thread and set
- a timeout to determine how long messages are kept around. It was
- always possible to implement this in the application with a custom
- log handler of course, this just provides this functionality as part
- of GStreamer.
-
-- 'fakevideosink is a null sink for video data that advertises
- video-specific metas ane behaves like a video sink. See above for
- more details.
+- A new helper base class for source elements, GstGLBaseSrc is
+ provided to ease writing source elements producing OpenGL video
+ frames.
+
+- Support for some more 12-bit and 16-bit video formats (Y412_LE,
+ Y412_BE, Y212_LE, Y212_BE, P012_LE, P012_BE, P016, NV16, NV61) was
+ added to glcolorconvert.
+
+- glupload can now import dma-buf’s into external-oes textures.
+
+- A new display type for EGLDevice-based systems was added. It is
+ currently opt-in by using either the GST_GL_PLATFORM=egl-device
+ environment variable or manual construction
+ (gst_gl_display_egl_device_new*()) due to compatibility issues with
+ some platforms.
+
+- Support was added for WinRT/UWP using the ANGLE project for running
+ OpenGL-based pipelines within a UWP application.
-- gst_util_dump_buffer() prints the content of a buffer to stdout.
-
-- gst_pad_link_get_name() and gst_state_change_get_name() print pad
- link return values and state change transition values as strings.
-
-- The LATENCY TRACER has seen a few improvements: trace records now
- contain timestamps which is useful to plot things over time, and
- downstream synchronisation time is now excluded from the measured
- values.
-
-- Miniobject refcount tracing and logging was not entirley
- thread-safe, there were duplicates or missing entries at times. This
- has now been made reliable.
-
-- The netsim element, which can be used to simulate network jitter,
- packet reordering and packet loss, received new features and
- improvements: it can now also simulate network congestion using a
- token bucket algorithm. This can be enabled via the "max-kbps"
- property. Packet reordering can be disabled now via the
- "allow-reordering" property: Reordering of packets is not very
- common in networks, and the delay functions will always introduce
- reordering if delay > packet-spacing, so by setting
- "allow-reordering" to FALSE you guarantee that the packets are in
- order, while at the same time introducing delay/jitter to them. By
- using the new "delay-distribution" property the user can control how
- the delay applied to delayed packets is distributed: This is either
- the uniform distribution (as before) or the normal distribution; in
- addition there is also the gamma distribution which simulates the
- delay on wifi networks better.
+- Various elements now support changing the GstGLDisplay to be used at
+ runtime in simple cases. This is primarily helpful for changing or
+ adding an OpenGL-based video sink that must share an OpenGL context
+ with an external source to an already running pipeline.
+GStreamer Vulkan integration
+
+- There is now a GStreamer Vulkan library to provide integration
+ points and helpers with applications and external GStreamer Vulkan
+ based elements. The structure of the library is modelled similarly
+ to the already existing GStreamer OpenGL library. Please note that
+ the API is still unstable and may change in future releases,
+ particularly around memory handling. The GStreamer Vulkan library
+ contains objects for sharing the vkInstance, vkDevice, vkQueue,
+ vkImage, VkMemory, etc with other elements and/or the application as
+ well as some helper objects for using Vulkan in an application or
+ element.
+
+- Added support for building and running on/for the Android and
+ Windows systems to complement the existing XCB, Wayland, MacOS, and
+ iOS implementations.
+
+- XCB gained support for mouse/keyboard events using the GstNavigation
+ API.
+
+- New vulkancolorconvert element for converting between color formats.
+ vulkancolorconvert can currently convert to/from all 8-bit RGBA
+ formats as well as 8-bit RGBA formats to/from the YUV formats AYUV,
+ NV12, and YUY2.
+
+- New vulkanviewconvert element for converting between stereo view
+ layouts. vulkanviewconvert can currently convert between all of the
+ single memory formats (side-by-side, top-bottom, column-interleaved,
+ row-interleaved, checkerboard, left, right, mono).
+
+- New vulkanimageidentity element for a blit from the input vulkan
+ image/s to a new vulkan image/s.
+
+- The vulkansink element can now scale the input image to the output
+ window/surface size where that information is available.
+
+- The vulkanupload element can now configure a transfer from system
+ memory to VulkanImage-based memory. Previously, this required two
+ vulkanupload elements.
+
+Tracing framework and debugging improvements
+
+- gst_tracing_get_active_tracers() returns a list of active tracer
+ objects. This can be used to interact with tracers at runtime using
+ GObject API such as action signals. This has been implemented in the
+ leaks tracer for snapshotting and retrieving leaked/active objects
+ at runtime.
+
+- The leaks tracer can now be interacted with programmatically at
+ runtime via GObject action signals:
+
+ - get-live-object returns a list of live (allocated) traced
+ objects
+ - log-live-objects logs a list of live objects into the debug log.
+ This is the same as sending the SIGUSR1 signal on unix systems,
+ but works on all operating systems including Windows.
+ - activity-start-tracking, activity-get-checkpoint,
+ activity-log-checkpoint, activity-stop-tracking: add support for
+ tracking and checkpointing objects, similar to what was
+ previously available via SIGUSR2 on unix systems, but works on
+ all operating systems including Windows.
+
+- various GStreamer gdb debug helper improvements:
+
+ - new ‘gst-pipeline-tree’ command
+ - more gdb helper functions: gst_element_pad(), gst_pipeline() and
+ gst_bin_get()
+ - support for queries and buffers
+ - print more info for segment events, print event seqnums, object
+ pointers and structures
+ - improve gst-print command to show more pad and element
+ information
Tools
-- gst-inspect-1.0 now prints pad properties for elements that have pad
- subclasses with special properties, such as compositor or
- audiomixer. This only works for elements that use the newly-added
- GstPadTemplate API API or the
- gst_element_class_add_static_pad_template_with_gtype() convenience
- function to tell GStreamer about the special pad subclass.
+gst-launch-1.0
-- gst-launch-1.0 now generates a gstreamer pipeline diagram (.dot
- file) whenever SIGHUP is sent to it on Linux/*nix systems.
+- now prints the pipeline position and duration if available when the
+ pipeline is advancing. This is hopefully more user-friendly and
+ gives visual feedback on the terminal that the pipeline is actually
+ up and running. This can be disabled with the --no-position command
+ line option.
-- gst-discoverer-1.0 can now analyse live streams such as rtsp:// URIs
+- the parse-launch pipeline syntax now has support for presets:
+ use@preset=<preset-name>" after an element to load a preset.
+gst-inspect-1.0
-GStreamer RTSP server
+- new --color command line option to force coloured output even if not
+ connected to a tty
-- Initial support for RTSP protocol version 2.0 was added, which is to
- the best of our knowledge the first RTSP 2.0 implementation ever!
+gst-tester-1.0 (new)
-- ONVIF audio backchannel support. This is an extension specified by
- ONVIF that allows RTSP clients (e.g. a control room operator) to
- send audio back to the RTSP server (e.g. an IP camera).
- Theoretically this could have been done also by using the RECORD
- method of the RTSP protocol, but ONVIF chose not to do that, so the
- backchannel is set up alongside the other streams. Format
- negotiation needs to be done out of band, if needed. Use the new
- ONVIF-specific subclasses GstRTSPOnvifServer and
- GstRTSPOnvifMediaFactory to enable this functionality.
+- gst-tester-1.0 is a new tool for plugin developers to launch
+ .validatetest files with TAP compatible output, meaning it can
+ easily and cleanly be integrated with the meson test harness. It
+ allows you to use gst-validate (from the gst-devtools module) to
+ write integration tests in any GStreamer repository whilst keeping
+ the tests as close as possible to the code. The tool transparently
+ handles gst-validate being installed or not: if it is not installed
+ those integration tests will simply be skipped.
-- The internal server streaming pipeline is now dynamically
- reconfigured on PLAY based on the transports needed. This means that
- the server no longer adds the pipeline plumbing for all possible
- transports from the start, but only if needed as needed. This
- improves performance and memory footprint.
+gst-play-1.0
-- rtspclientsink has gained an "accept-certificate" signal for
- manually checking a TLS certificate for validity.
+- interactive keyboard controls now also work on Windows
-- Fix keep-alive/timeout issue for certain clients using TCP
- interleave as transport who don't do keep-alive via some other
- method such as periodic RTSP OPTION requests. We now put netaddress
- metas on the packets from the TCP interleaved stream, so can map
- RTCP packets to the right stream in the server and can handle them
- properly.
+gst-transcoder-1.0 (new)
-- Language bindings improvements: in general there were quite a few
- improvements in the gobject-introspection annotations, but we also
- extended the permissions API which was not usable from bindings
- before.
+- gst-transcoder-1.0 is a new command line tool to transcode one URI
+ into another URI based on the specified encoding profile using the
+ new GstTranscoder API (see above).
-- Fix corner case issue where the wrong mount point was found when
- there were multiple mount points with a common prefix.
+GStreamer RTSP server
+- Fix issue where the first few packets (i.e. keyframes) could
+ sometimes be dropped if the rtsp media pipeline had a live input.
+ This was a regression from GStreamer 1.14. There are more fixes
+ pending for that which will hopefully land in 1.18.1.
+
+- Fix backpressure handling when sending data in TCP interleave mode
+ where RTSP requests and responses and RTP/RTCP packets flow over the
+ same RTSP TCP connection: The previous implementation would at some
+ point stop sending data to other clients when a single client
+ stopped consuming data or did not consume data fast enough. This
+ obviously created problems for shared media, where the same stream
+ from a single producer pipeline is sent to multiple clients. Instead
+ we now manage a backlog in the server’s stream-transport component
+ and remove slow clients once this backlog exceeds a maximum duration
+ (which is currently hardcoded).
+
+- Onvif Streaming Specification trick modes support (see section at
+ the beginning)
+
+- Scale/Speed header support: Speed will deliver the data at the
+ requested speed, which means increasing the data bandwidth for
+ speeds > 1.0. Scale will attempt to do the same without affecting
+ the overall bandwidth requirement vis-a-vis normal playback speed
+ (e.g. it might drop data for fast-forward playback).
+
+- rtspclientsink: send buffer lists in one go for better performance
GStreamer VAAPI
-- Improve DMABuf's usage, both upstream and dowstream, and
- memory:DMABuf caps feature is also negotiated when the dmabuf-based
- buffer cannot be mapped onto user-space.
+- A lot of work was done adding support for media-driver (iHD), the
+ new VAAPI driver for Intel, mostly for Gen9 onwards.
+
+- Available color formats and frame sizes are now detected at run-time
+ according to the context configuration.
+
+- Gallium drivers have been re-enabled in the allowed drivers list
+
+- Improved the mapping between VA formats and GStreamer formats by
+ generating a mapping table at run-time since even among different
+ drivers the mapping might be different, particularly for RGB with
+ little endianness.
+
+- The experimental Flexible Encoding Infrastructure (FEI) elements
+ have been removed since they were not really actively maintained or
+ tested.
-- VA initialization was fixed when it is used in headless systems.
+- Enhanced the juggling of DMABuf buffers and VASurface metas
-- VA display sharing, through GstContext, among the pipeline, has been
- improved, adding the possibility to the application share its VA
- display (external display) via gst.vaapi.app.Display context.
+- New vaapioverlay element: a compositor element using VA VPP blend
+ capabilities to accelerate overlaying and compositing. Example
+ pipeline:
-- VA display cache was removed.
+ gst-launch-1.0 -vf videotestsrc ! vaapipostproc ! tee name=testsrc ! queue \
+ ! vaapioverlay sink_1::xpos=300 sink_1::alpha=0.75 name=overlay ! vaapisink \
+ testsrc. ! queue ! overlay.
-- libva's log messages are now redirected into the GStreamer log
- handler.
+vaapipostproc
-- Decoders improved their upstream re-negotiation by avoiding to
- re-instantiate the internal decoder if stream caps are compatible
- with the previous one.
+- added video-orientation support, supporting frame mirroring and
+ rotation
-- When downstream doesn't support GstVideoMeta and the decoded frames
- don't have standard strides, they are copied onto system
- memory-based buffers.
+- added cropping support, either via properties (crop-left,
+ crop-right, crop-bottom and crop-top) or buffer meta.
-- H.264 decoder has a low-latency property, for live streams which
- doesn't conform the H.264 specification but still it is required to
- push the frames to downstream as soon as possible.
+- new skin-tone-enhancenment-level property which is the iHD
+ replacement of the i965 driver’s sink-tone-level. Both are
+ incompatible with each other, so both were kept.
-- As part of the Google Summer of Code 2017 the H.264 decoder drops
- MVC and SVC frames when base-only property is enabled.
+- handle video colorimetry
-- Added support for libva-2.0 (VA-API 1.0).
+- support HDR10 tone mapping
-- H.264 and H.265 encoders handle Region-Of-Interest metas by adding a
- delta-qp for every rectangle within the frame specified by those
- metas.
+vaapisink
-- Encoders for H.264 and H.265 set the media profile by the downstream
- caps.
+- resurrected wayland backend for non-weston compositors by extracting
+ the DMABuf from the VASurface and rendering it.
-- H.264 encoder inserts an AU delimiter for each encoded frame when
- aud property is enabled (it is only available for certain drivers
- and platforms).
+- merged the video overlay API for wayland. Now applications can
+ define the “window” to render on.
-- H.264 encoder supports for P and B hierarchical prediction modes.
+- demoted the vaapisink element to secondary rank since libva
+ considers rendering as a second-class feature.
-- All encoders handles a quality-level property, which is a number
- from 1 to 8, where a lower number means higher quality, but slower
- processing, and vice-versa.
+VAAPI Encoders
-- VP8 and VP9 encoders support constant bit-rate mode (CBR).
+- new common target-percentage property which is the desired target
+ percentage of bitrate for variable rate control.
-- VP8, VP9 and H.265 encoders support variable bit-rate mode (VBR).
+- encoders now extract their caps from the driver at registration
+ time.
-- Resurrected GstGLUploadTextureMeta handling for EGL backends.
+- vaapivp9enc: added support for low power mode and support for
+ profile 2 (profile 0 by default)
-- H.265 encoder can configure its number of reference frames via the
- refs property.
+- vaapih264enc: new max-qp property that sets the maximum quantization
+ value. Support for ICQ and QBVR bitrate control mode, adding a
+ quality-factor property for these modes. Support baseline profile as
+ constrained-baseline
-- Add H.264 encoder mbbrc property, which controls the macro-block
- bitrate as auto, on or off.
+- vaapih265enc:
-- Add H.264 encoder temporal-levels property, to select the number of
- temporal levels to be included.
+ - support for main-444 and main-12 encoding profiles.
+ - new max-qp property that sets the maximum quantization value.
+ - support for ICQ and QBVR bitrate control mode, adding a
+ quality-factor property for these modes.
+ - handle SCC profiles.
+ - num-tile-cols and num-tile-row properties to specify the number
+ of tiles to use.
+ - the low-delay-b property was deprecated and is now determined
+ automatically.
+ - improved profile selection through caps.
-- Add to H.264 and H.265 encoders the properties qp-ip and qp-ib, to
- handle the QP (quality parameter) difference between the I and P
- frames, and the I and B frames, respectively.
+VAAPI Decoders
-- vaapisink was demoted to marginal rank on Wayland because COGL
- cannot display YUV surfaces.
+- Decoder surfaces are not bound to their context any longer and can
+ thus be created and used dynamically, removing the deadlock
+ headache.
+- Reverse playback is now fluid
+
+- Forward Region-of-Interest (ROI) metas downstream
+
+- GLTextureUploadMeta uses DMABuf when GEM is not available. Now
+ Gallium drivers can use this meta for rendering with EGL.
+
+- vaapivp9dec: support for 4:2:2 and 4:4:4 chroma type streams
+
+- vaapih265dec: skip all pictures prior to the first I-frame. Enable
+ passing range extension flags to the driver. Handle SCC profiles.
+
+- vaapijpegdec: support for 4:0:0, 4:1:1, 4:2:2 and 4:4:4 chroma types
+ pictures
+
+- vaapih264dec: handle baseline streams as constrained-baseline if
+ possible and make it more tolerant when encountering unknown NALs
+
+GStreamer OMX
+
+- omxvideoenc: use new video encoder subframe API to push out slices
+ as soon as they’re ready
+
+- omxh264enc, omxh265enc: negotiate subframe mode via caps. To enable
+ it, force downstream caps to video/x-h264,alignment=nal or
+ video/x-h265,alignment=nal.
+
+- omxh264enc: Add ref-frames property
+
+- Zynq ultrascale+ specific video encoder/decoder improvements:
+
+ - GRAY8 format support
+ - support for alternate fields interlacing mode
+ - video encoder: look-ahead, long-term-ref, and long-term-freq
+ properties
GStreamer Editing Services and NLE
-- Handle crossfade in complex scenarios by using the new
- compositorpad::crossfade-ratio property
+- Added nested timelines and subproject support so that GES projects
+ can be used as clips, potentially serializing nested projects in the
+ main file or referencing external project files.
-- Add API allowing to stop using proxies for clips in the timeline
+- Implemented an OpenTimelineIO GES formatter. This means GES and
+ GStreamer can now load and save projects in all the formats
+ supported by otio.
-- Allow management of none square pixel aspect ratios by allowing
- application to deal with them in the way they want
+- Implemented a GESMarkerList object which allow setting timed
+ metadata on any GES object.
-- Misc fixes around the timeline editing API
+- Fixed audio rendering issues during clip transition by ensuring that
+ a single segment is pushed into encoders.
+- The GESUriClipAsset API is now MT safe.
-GStreamer validate
+- Added ges_meta_container_register_static_meta() to allow fixing a
+ type for a specific metadata without actually setting a value.
-- Handle running scenarios on live pipelines (in the "content sense",
- not the GStreamer one)
+- The framepositioner element now handles resizing the project and
+ keeps the same positioning when the aspect ratio is not changed .
-- Implement RTSP support with a basic server based on gst-rtsp-server,
- and add RTSP 1.0 and 2.0 integration tests
+- Reworked the documentation, making it more comprehensive and much
+ more detailed.
-- Implement a plugin that allows users to implement configurable
- tests. It currently can check if a particular element is added a
- configurable number of time in the pipeline. In the future that
- plugin should allow us to implement specific tests of any kind in a
- descriptive way
+- Added APIs to retrieve natural size and framerate of a clip (for
+ example in the case of URIClip it is the framerate/size of the
+ underlying file).
-- Add a verbosity configuration which behaves in a similare way as the
- gst-launch-1.0 verbose flags allowing the informations to be
- outputed on any running pipeline when enabling GstValidate.
+- ges_container_edit() is now deprecated and GESTimelineElement gained
+ the ges_timeline_element_edit() method so the editing API is now
+ usable from any element in the timeline.
-- Misc optimization in the launcher, making the tests run much faster.
+- GESProject::loading was added so applications can be notified about
+ when a new timeline starts loading.
+- Implemented the GstStream API in GESTimeline.
-GStreamer C# bindings
+- Added a way to add a timeoverlay inside the test source (potentially
+ with timecodes).
-- Port to the meson build system, autotools support has been removed
+- Added APIs to convert times to frame numbers and vice versa:
-- Use a new GlibSharp version, set as a meson subproject
+ - ges_timeline_get_frame_time()
-- Update wrapped API to GStreamer 1.14
+ - ges_timeline_get_frame_at()
-- Removed the need for "glue" code
+ - ges_clip_asset_get_frame_time()
-- Provide a nuget
+ - ges_clip_get_timeline_time_from_source_frame()
-- Misc API fixes
+ Quite a few validate tests have been implemented to check the
+ behavior for various demuxer/codec formats
+- Added ges_layer_set_active_for_tracks() which allows muting layers
+ for the specified tracks
-Build and Dependencies
+- Deprecated GESImageSource and GESMultiFileSource now that we have
+ imagesequencesrc which handles the imagesequence “protocol”
-- the new WebRTC support in gst-plugins-bad depends on the GStreamer
- elements that ship as part of libnice, and libnice version 1.1.14 is
- required. Also the dtls and srtp plugins.
+- Stopped exposing ‘deinterlacing’ children properties for clip types
+ where they do not make sense.
-- gst-plugins-bad no longer depends on the libschroedinger Dirac codec
- library.
+- Added support for simple time remapping effects
-- The srtp plugin can now also be built against libsrtp2.
+GStreamer validate
-- some plugins and libraries have moved between modules, see the
- _Plugin and_ _library moves_ section above, and their respective
- dependencies have moved with them of course, e.g. the GStreamer
- OpenGL integration support library and plugin is now in
- gst-plugins-base, and mpg123, LAME and twoLAME based audio decoder
- and encoder plugins are now in gst-plugins-good.
+- Introduced the concept of “Test files” allowing to implement “all
+ included” test cases, meaning that inside the file the following can
+ be defined:
-- Unify static and dynamic plugin interface and remove plugin specific
- static build option: Static and dynamic plugins now have the same
- interface. The standard --enable-static/--enable-shared toggle is
- sufficient. This allows building static and shared plugins from the
- same object files, instead of having to build everything twice.
+ - The application arguments
+ - The validate configurations
+ - The validate scenario
-- The default plugin entry point has changed. This will only affect
- plugins that are recompiled against new GStreamer headers. Binary
- plugins using the old entry point will continue to work. However,
- plugins that are recompiled must have matching plugin names in
- GST_PLUGIN_DEFINE and filenames, as the plugin entry point for
- shared plugins is now deduced from the plugin filename. This means
- you can no longer have a plugin called foo living in a file called
- libfoobar.so or such, the plugin filename needs to match. This might
- cause problems with some external third party plugin modules when
- they get rebuilt against GStreamer 1.14.
+ This replaces the previous big dictionary file in
+ gst-validate-launcher to implement specific test cases.
+ We set several variables inside the files (as well as inside
+ scenarios and config files) to make them relocatable.
-Note to packagers and distributors
+ The file format has been enhanced so it is easier to read and write,
+ for example line ending with a coma or (curly) brackets can now be
+ used as continuation marker so you do not need to add \ at the end
+ of lines to write a structure on several lines.
-A number of libraries, APIs and plugins moved between modules and/or
-libraries in different modules between version 1.12.x and 1.14.x, see
-the _Plugin and_ _library moves_ section above. Some APIs have seen
-minor ABI changes in the course of moving them into the stable APIs
-section.
+- Support the imagesequence “protocol” and added integration tests for
+ it.
-This means that you should try to ensure that all major GStreamer
-modules are synced to the same major version (1.12 or 1.13/1.14) and can
-only be upgraded in lockstep, so that your users never end up with a mix
-of major versions on their system at the same time, as this may cause
-breakages.
+- Added action types to allow the scenario to run the Test Clock for
+ better reproducibility of tests.
-Also, plugins compiled against >= 1.14 headers will not load with
-GStreamer <= 1.12 owing to a new plugin entry point (but plugin binaries
-built against older GStreamer versions will continue to load with newer
-versions of GStreamer of course).
+- Support generating tests to check that seeking is frame accurate
+ (base on ssim).
-There is also a small structure size related ABI breakage introduced in
-the gst-plugins-bad codecparsers library between version 1.13.90 and
-1.13.91. This should "only" affect gstreamer-vaapi, so anyone who ships
-the release candidates is advised to upgrade those two modules at the
-same time.
+- Added ways to record buffers checksum (in different ways) in the
+ validateflow module.
+- Added vp9 encoding tests.
-Platform-specific improvements
+- Enhanced seeking action types implementation to allow support for
+ segment seeks.
-Android
+- Output improvements:
-- ahcsrc (Android camera source) does autofocus now
+ - Logs are now in markdown formats (and bat is used to dump them
+ if available).
+ - File format issues in scenarios/configs/tests files are nicely
+ reported with the line numbers now.
-macOS and iOS
+GStreamer Python Bindings
-- this section will be filled in shortly {FIXME!}
+- Python 2.x is no longer supported
-Windows
+- Support mapping buffers without any memcpy:
-- The GStreamer wasapi plugin was rewritten and should not only be
- usable now, but in top shape and suitable for low-latency use cases.
- The Windows Audio Session API (WASAPI) is Microsoft's most modern
- method for talking with audio devices, and now that the wasapi
- plugin is up to scratch it is preferred over the directsound plugin.
- The ranks of the wasapisink and wasapisrc elements have been updated
- to reflect this. Further improvements include:
+ - Added a ContextManager to make the API more pythonic
-- support for more than 2 channels
+ with buf.map(Gst.MapFlags.READ | Gst.MapFlags.WRITE) as info:
+ info.data[42] = 0
-- a new "low-latency" property to enable low-latency operation (which
- should always be safe to enable)
+- Added high-level helper API for constructing pipelines:
-- support for the AudioClient3 API which is only available on Windows
- 10: in wasapisink this will be used automatically if available; in
- wasapisrc it will have to be enabled explicitly via the
- "use-audioclient3" property, as capturing audio with low latency and
- without glitches seems to require setting the realtime priority of
- the entire pipeline to "critical", which cannot be done from inside
- the element, but has to be done in the application.
+ - Gst.Bin.make_and_add(factory_name, instance_name=None)
+ - Gst.Element.link_many(element, ...)
-- set realtime thread priority to avoid glitches
+GStreamer C# Bindings
-- allow opening devices in exclusive mode, which provides much lower
- latency compared to shared mode where WASAPI's engine period is
- 10ms. This can be activated via the "exclusive" property.
+- Bind gst_buffer_new_wrapped() manually to fix memory handling.
-- There are now GstDeviceProvider implementations for the wasapi and
- directsound plugins, so it's now possible to discover both audio
- sources and audio sinks on Windows via the GstDeviceMonitor API
+- Fix gst_promise_new_with_change_func() where bindgen didn’t properly
+ detect the func as a closure.
-- debug log timestamps are now higher granularity owing to
- g_get_monotonic_time() now being used as fallback in
- gst_utils_get_timestamp(). Before that, there would sometimes be
- 10-20 lines of debug log output sporting the same timestamp.
+- Declare GstVideoOverlayComposition and GstVideoOverlayRectangle as
+ opaque type and subclasses of Gst.MiniObject. This changes the API
+ but without this all usage will cause memory corruption or simply
+ not work.
+- on Windows, look for gstreamer, glib and gobject DLLs using the MSVC
+ naming convention (i.e. gstvideo-1.0-0.dll instead of
+ libgstvideo-1.0-0.dll).
-Contributors
+ The names of these DLLs have to be hardcoded in the bindings, and
+ most C# users will probably be using the Microsoft toolchain anyway.
-Aaron Boxer, Adrián Pardini, Adrien SCH, Akinobu Mita, Alban Bedel,
-Alessandro Decina, Alex Ashley, Alicia Boya García, Alistair Buxton,
-Alvaro Margulis, Anders Jonsson, Andreas Frisch, Andrejs Vasiljevs,
-Andrew Bott, Antoine Jacoutot, Antonio Ospite, Antoni Silvestre, Anton
-Obzhirov, Anuj Jaiswal, Arjen Veenhuizen, Arnaud Bonatti, Arun Raghavan,
-Ashish Kumar, Aurélien Zanelli, Ayaka, Branislav Katreniak, Branko
-Subasic, Brion Vibber, Carlos Rafael Giani, Cassandra Rommel, Chris
-Bass, Chris Paulson-Ellis, Christoph Reiter, Claudio Saavedra, Clemens
-Lang, Cyril Lashkevich, Daniel van Vugt, Dave Craig, Dave Johnstone,
-David Evans, David Schleef, Deepak Srivastava, Dimitrios Katsaros,
-Dmitry Zhadinets, Dongil Park, Dustin Spicuzza, Eduard Sinelnikov,
-Edward Hervey, Enrico Jorns, Eunhae Choi, Ezequiel Garcia, fengalin,
-Filippo Argiolas, Florent Thiéry, Florian Zwoch, Francisco Velazquez,
-François Laignel, fvanzile, George Kiagiadakis, Georg Lippitsch, Graham
-Leggett, Guillaume Desmottes, Gurkirpal Singh, Gwang Yoon Hwang, Gwenole
-Beauchesne, Haakon Sporsheim, Haihua Hu, Håvard Graff, Heekyoung Seo,
-Heinrich Fink, Holger Kaelberer, Hoonhee Lee, Hosang Lee, Hyunjun Ko,
-Ian Jamison, James Stevenson, Jan Alexander Steffens (heftig), Jan
-Schmidt, Jason Lin, Jens Georg, Jeremy Hiatt, Jérôme Laheurte, Jimmy
-Ohn, Jochen Henneberg, John Ludwig, John Nikolaides, Jonathan Karlsson,
-Josep Torra, Juan Navarro, Juan Pablo Ugarte, Julien Isorce, Jun Xie,
-Jussi Kukkonen, Justin Kim, Lasse Laursen, Lubosz Sarnecki, Luc
-Deschenaux, Luis de Bethencourt, Marcin Lewandowski, Mario Alfredo
-Carrillo Arevalo, Mark Nauwelaerts, Martin Kelly, Matej Knopp, Mathieu
-Duponchelle, Matteo Valdina, Matt Fischer, Matthew Waters, Matthieu
-Bouron, Matthieu Crapet, Matt Staples, Michael Catanzaro, Michael
-Olbrich, Michael Shigorin, Michael Tretter, Michał Dębski, Michał Górny,
-Michele Dionisio, Miguel París, Mikhail Fludkov, Munez, Nael Ouedraogo,
-Neos3452, Nicholas Panayis, Nick Kallen, Nicola Murino, Nicolas
-Dechesne, Nicolas Dufresne, Nirbheek Chauhan, Ognyan Tonchev, Ole André
-Vadla Ravnås, Oleksij Rempel, Olivier Crête, Omar Akkila, Orestis
-Floros, Patricia Muscalu, Patrick Radizi, Paul Kim, Per-Erik Brodin,
-Peter Seiderer, Philip Craig, Philippe Normand, Philippe Renon, Philipp
-Zabel, Pierre Pouzol, Piotr Drąg, Ponnam Srinivas, Pratheesh Gangadhar,
-Raimo Järvi, Ramprakash Jelari, Ravi Kiran K N, Reynaldo H. Verdejo
-Pinochet, Rico Tzschichholz, Robert Rosengren, Roland Peffer, Руслан
-Ижбулатов, Sam Hurst, Sam Thursfield, Sangkyu Park, Sanjay NM, Satya
-Prakash Gupta, Scott D Phillips, Sean DuBois, Sebastian Cote, Sebastian
-Dröge, Sebastian Rasmussen, Sejun Park, Sergey Borovkov, Seungha Yang,
-Shakin Chou, Shinya Saito, Simon Himmelbauer, Sky Juan, Song Bing,
-Sreerenj Balachandran, Stefan Kost, Stefan Popa, Stefan Sauer, Stian
-Selnes, Thiago Santos, Thibault Saunier, Thijs Vermeir, Tim Allen,
-Tim-Philipp Müller, Ting-Wei Lan, Tomas Rataj, Tom Bailey, Tonu Jaansoo,
-U. Artie Eoff, Umang Jain, Ursula Maplehurst, VaL Doroshchuk, Vasilis
-Liaskovitis, Víctor Manuel Jáquez Leal, vijay, Vincent Penquerc'h,
-Vineeth T M, Vivia Nikolaidou, Wang Xin-yu (王昕宇), Wei Feng, Wim
-Taymans, Wonchul Lee, Xabier Rodriguez Calvar, Xavier Claessens,
-XuGuangxin, Yasushi SHOJI, Yi A Wang, Youness Alaoui,
-
-... and many others who have contributed bug reports, translations, sent
-suggestions or helped testing.
+ This means that the MSVC compiler is now required to build the
+ bindings, MingW will no longer work out of the box.
+GStreamer Rust Bindings and Rust Plugins
-Bugs fixed in 1.14
+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 new GStreamer 1.18 API, so there’s
+absolutely no excuse why your next GStreamer application can’t be
+written in Rust anymore.
-More than 800 bugs have been fixed during the development of 1.14.
+gst-plugins-rs, the module containing GStreamer plugins written in Rust,
+has also seen lots of activity with many new elements and plugins.
-This list does not include issues that have been cherry-picked into the
-stable 1.12 branch and fixed there as well, all fixes that ended up in
-the 1.12 branch are also included in 1.14.
+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.
-This list also does not include issues that have been fixed without a
-bug report in bugzilla, so the actual number of fixes is much higher.
+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
-Stable 1.14 branch
+Rust video plugins
-After the 1.14.0 release there will be several 1.14.x bug-fix releases
-which will contain bug fixes which have been deemed suitable for a
-stable branch, but no new features or intrusive changes will be added to
-a bug-fix release usually. The 1.14.x bug-fix releases will be made from
-the git 1.14 branch, which is a stable branch.
+- cdgdec/cdgparse: Decoder and parser for the CD+G video codec based
+ on a pure-Rust CD+G implementation, used for example by karaoke CDs
+- cea608overlay: CEA-608 Closed Captions overlay element
+- cea608tott: CEA-608 Closed Captions to timed-text (e.g. VTT or SRT
+ subtitles) converter
+- tttocea608: CEA-608 Closed Captions from timed-text converter
+- mccenc/mccparse: MacCaption Closed Caption format encoder and parser
+- sccenc/sccparse: Scenarist Closed Caption format encoder and parser
+- dav1dec: AV1 video decoder based on the dav1d decoder implementation
+ by the VLC project
+- rav1enc: AV1 video encoder based on the fast and pure-Rust rav1e
+ encoder implementation
+- rsflvdemux: Alternative to the flvdemux FLV demuxer element from
+ gst-plugins-good, not feature-equivalent yet
+- rsgifenc/rspngenc: GIF/PNG encoder elements based on the pure-Rust
+ implementations by the image-rs project
+
+Rust text plugins
+
+- textwrap: Element for line-wrapping timed text (e.g. subtitles) for
+ better screen-fitting, including hyphenation support for some
+ languages
+
+Rust network plugins
+
+- reqwesthttpsrc: HTTP(S) source element based on the Rust
+ reqwest/hyper HTTP implementations and almost feature-equivalent
+ with the main GStreamer HTTP source souphttpsrc
+- s3src/s3sink: Source/sink element for the Amazon S3 cloud storage
+- awstranscriber: Live audio to timed text transcription element using
+ the Amazon AWS Transcribe API
+
+Generic Rust plugins
+
+- sodiumencrypter/sodiumdecrypter: Encryption/decryption element based
+ on libsodium/NaCl
+- togglerecord: Recording element that allows to pause/resume
+ recordings easily and considers keyframe boundaries
+- fallbackswitch/fallbacksrc: Elements for handling potentially
+ failing (network) sources, restarting them on errors/timeout and
+ showing a fallback stream instead
+- threadshare: Set of elements that provide alternatives for various
+ existing GStreamer elements but allow to share the streaming threads
+ between each other to reduce the number of threads
+- rsfilesrc/rsfilesink: File source/sink elements as replacements for
+ the existing filesrc/filesink elements
+
+Build and Dependencies
+
+- The Autotools build system has finally been removed in favour of the
+ Meson build system. Developers who currently use gst-uninstalled
+ should move to gst-build.
+
+- API and plugin documentation are no longer built with gtk_doc. The
+ gtk_doc documentation has been removed in favour of a new unified
+ documentation module built with hotdoc (also see “Documentation
+ improvements” section below). Distributors should use the
+ documentation release tarball instead of trying to package hotdoc
+ and building the documentation from scratch.
+
+- gst-plugins-bad now includes an internal copy of libusrsctp, as
+ there are problems in usrsctp with global shared state, lack of API
+ stability guarantees, and the absence of any kind of release
+ process. We also can’t rely on distros shipping a version with the
+ fixes we need. Both firefox and Chrome bundle their own copies too.
+ It is still possible to build against an external copy of usrsctp if
+ so desired.
+
+- nvcodec no longer needs the NVIDIA NVDEC/NVENC SDKs available at
+ build time, only at runtime. This allows distributions to ship this
+ plugin by default and it will just start to work when the required
+ run-time SDK libraries are installed by the user, without users
+ needing to build and install the plugin from source.
+
+- the gst-editing-services tarball is now named gst-editing-services
+ for consistency (used to be gstreamer-editing-services).
+
+- the gst-validate tarball has been superseded by the gst-devtools
+ tarball for consistency with the git module name.
+
+gst-build
+
+gst-build is a meta-module and serves primarily as our uninstalled
+development environment. It makes it easy to build most of GStreamer,
+but unlike Cerbero it only comes with a limited number of external
+dependencies that can be built as subprojects if they are not found on
+the system.
+
+gst-build is based on Meson and replaces the old autotools
+gst-uninstalled script.
+
+- The ‘uninstalled’ target has been renamed to ‘devenv’
+
+- Experimental gstreamer-full library containing all built plugins and
+ their deps when building with -Ddefault_library=static. A monolithic
+ library is easier to distribute, and may be required in some
+ environments. GStreamer core, GLib and GObject are always included,
+ but external dependencies are still dynamically linked. The
+ gst-full-libraries meson option allows adding other GStreamer
+ libraries to the gstreamer-full build. This is an experiment for now
+ and its behaviour or API may still change in future releases.
+
+- Add glib-networking as a subproject when glib is a subproject and
+ load gio modules in the devenv, tls option control whether to use
+ openssl or gnutls.
+
+- git-worktree: Allow multiple worktrees for subproject branches
+
+- Guard against meson being run from inside the uninstalled devenv, as
+ this might have unexpected consequences.
+
+- our ffmpeg and x264 meson ports have been updated to the latest
+ stable version (you might need to update the subprojects checkout
+ manually though, or just remove the checkouts so meson checks out
+ the latest version again; improvements for this are pending in
+ meson, but not merged yet).
+
+Cerbero
+
+Cerbero is a meta build system used to build GStreamer plus dependencies
+on platforms where dependencies are not readily available, such as
+Windows, Android, iOS and macOS.
+
+General improvements
+
+- Recipe build steps are done in parallel wherever possible. This
+ leads to massive improvements in overall build time.
+- Several recipes were ported to Meson, which improved build times
+- Moved from using both GnuTLS and OpenSSL to only OpenSSL
+- Moved from yasm to nasm for all assembly compilation
+- Support zsh when running the cerbero shell command
+- Numerous version upgrades for dependencies
+- Default to xz for tarball binary packages. bz2 can be selected with
+ the --compress-method option to package.
+- Added boolean variant for controlling the optimization level:
+ -v optimization
+- Ship .pc pkgconfig files for all plugins in the binary packages
+- CMake and nasm will only be built by Cerbero if the system versions
+ are unusable
+- The nvcodec variant was removed and the nvcodec plugin is built by
+ default now (as it no longer requires the SDK to be installed at
+ build time, only at runtime)
+
+macOS / iOS
+
+- Minimum iOS SDK version bumped to 11.0
+- Minimum macOS SDK version bumped to 10.11
+- No longer need to manually add support for newer iOS SDK versions
+- Added Vulkan elements via MoltenVK
+- Build times were improved by code-signing all build tools
+- macOS framework ships all gstreamer libraries instead of an outdated
+ subset
+- Ship pkg-config in the macOS framework package
+- fontconfig: Fix EXC_BAD_ACCESS crash on iOS ARM64
+- Improved App Store compatibility by setting LC_VERSION_MIN_MACOSX,
+ fixing relocations, and improved bitcode support
+
+Windows
+
+- MinGW-GCC toolchain was updated to 8.2. It uses the Universal CRT
+ instead of MSVCRT which eliminates cross-CRT issues in the Visual
+ Studio build.
+- Require Windows 7 or newer for running binaries produced by Cerbero
+- Require Windows x86_64 for running Cerbero to build binary packages
+- Cerbero no longer uses C:/gstreamer/1.0 as a prefix when building.
+ That prefix is reserved for use by the MSI installers.
+- Several recipes can now be buit with Visual Studio instead of MinGW.
+ Ported to meson: opus, libsrtp, harfbuzz, cairo, openh264, libsoup,
+ libusrsctp. Existing build system: libvpx, openssl.
+- Support building using Visual Studio for 32-bit x86. Previously we
+ only supported building for 32-bit x86 using the MinGW toolchain.
+- Fixed annoying msgmerge popups in the middle of cerbero builds
+- Added configuration options vs_install_path and vs_install_version
+ for specifying custom search locations for older Visual Studio
+ versions that do not support vswhere. You can set these in
+ ~/.cerbero/cerbero.cbc where ~ is the MSYS homedir, not your Windows
+ homedir.
+- New Windows-specific plugins: d3d11, mediafoundation, wasapi2
+- Numerous compatibility and reliability fixes when running Cerbero on
+ Windows, especially non-English locales
+- proxy-libintl now exports the same symbols as gettext, which makes
+ it a drop-in replacement
+- New mapping variant for selecting the Visual Studio CRT to use:
+ -v vscrt=<value>. Valid values are md, mdd, and auto (default). A
+ separate prefix is used when building with either md (release) or
+ mdd (debug), and the outputted package will have +debug in the
+ filename. This variant is also used for selecting the correct Qt
+ libraries (debug vs release) to use when building with -v qt5 on
+ Windows.
+- Support cross-compile on Windows to Windows ARM64 and ARMv7
+- Support cross-compile on Windows to the Universal Windows Platform
+ (UWP). Only the subset of plugins that can be built entirely with
+ Visual Studio will be selected in this case. To do so, use the
+ config/cross-uwp-universal.cbc configuration, which will build
+ ARM64, x86, and x86_64 binaries linked to the release CRT, with
+ optimizations enabled, and debugging turned on. You can combine this
+ with -v vscrt=mdd to produce binaries linked to the debug CRT. You
+ can turn off optimizations with the -v nooptimization variant.
+
+Windows MSI installer
+
+- Require Windows 7 or newer for running GStreamer
+- Fixed some issues with shipping of pkg-config in the Windows
+ installers
+- Plugin PDB debug files are now shipped in the development package,
+ not the runtime package
+- Ship installers for 32-bit binaries built with Visual Studio
+- Ship debug and release “universal” (ARM64, X86, and X86_64) tarballs
+ built for the Universal Windows Platform
+- Windows MSI installers now install into separate prefixes when
+ building with MSVC and MinGW. Previously both would be installed
+ into C:/gstreamer/1.0/x86 or C:/gstreamer/1.0/x86_64. Now, the
+ installation prefixes are:
+
+ ----------------------------------------------------------------------------------------------------------------
+ Target Path Build options
+ --------------------------- ------------------------------------ -----------------------------------------------
+ MinGW 32-bit C:/gstreamer/1.0/mingw_x86 -c config/win32.cbc
+
+ MinGW 64-bit C:/gstreamer/1.0/mingw_x86_64 -c config/win64.cbc
+
+ MSVC 32-bit C:/gstreamer/1.0/msvc_x86 -c config/win32.cbc -v visualstudio
+
+ MSVC 64-bit C:/gstreamer/1.0/msvc_x86_64 -c config/win64.cbc -v visualstudio
+
+ MSVC 32-bit (debug) C:/gstreamer/1.0/msvc-debug_x86 -c config/win32.cbc -v visualstudio,vscrt=mdd
+
+ MSVC 64-bit (debug) C:/gstreamer/1.0/msvc-debug_x86_64 -c config/win64.cbc -v visualstudio,vscrt=mdd
+ ----------------------------------------------------------------------------------------------------------------
+
+Note: UWP binary packages are tarballs, not MSI installers.
+
+Linux
+
+- Support creating MSI installers using WiX when cross-compiling to
+ Windows
+- Support running cross-windows binaries with Wine when using the
+ shell and runit cerbero commands
+- Added bash-completion support inside the cerbero shell on Linux
+- Require a system-wide installation of openssl on Linux
+- Added variant -v vaapi to build gstreamer-vaapi and the new gstva
+ plugin
+- Debian packaging was disabled because it does not work. Help in
+ fixing this is appreciated.
+- Trimmed the list of packages needed for bootstrap on Linux
+
+Android
-1.14.0
+- Updated to NDK r21
+- Support Vulkan
+- Support Qt 5.14+ binary package layout
-1.14.0 was released on 19 March 2018.
+Platform-specific changes and improvements
-1.14.1
+Android
+
+- opensles: Remove hard-coded buffer-/latency-time values and allow
+ openslessink to handle 48kHz streams.
-The first 1.14 bug-fix release (1.14.1) is scheduled to be released
-around the end of March or beginning of April.
+- photography interface and camera source: Add additional settings
+ relevant to Android such as: Exposure mode property, extra colour
+ tone values (aqua, emboss, sketch, neon), extra scene modes
+ (backlight, flowers, AR, HDR), and missing virtual methods for
+ exposure mode, analog gain, lens focus, colour temperature, min &
+ max exposure time. Add new effects and scene modes to Camera
+ parameters.
-This release only contains bugfixes and it should be safe to update from
-1.14.0.
+macOS and iOS
+- vtdec can now output to Vulkan-backed memory for zerocopy support
+ with the Vulkan elements.
+
+Windows
+
+- d3d11videosink: new Direct3D11-based video sink with support for
+ HDR10 rendering if supported.
+
+- Hardware-accelerated video decoding on Windows via DXVA2 /
+ Direct3D11 using native Windows APIs rather than per-vendor SDKs
+ (like MSDK for Intel or NVCODEC for NVidia). Plus modern Direct3D11
+ integration rather than the almost 20-year old Direct3D9 from
+ Windows XP times used in d3dvideosink. Formats supported for
+ decoding are H.264, H.265, VP8, and VP9, and zero-copy operation
+ should be supported in combination with the new d3d11videosink. See
+ Seungha’s blog post “Windows DXVA2 (via Direct3D 11) Support in
+ GStreamer 1.17” for more details.
+
+- Microsoft Media Foundation plugin for hardware-accelerated video
+ encoding on Windows using native Windows APIs rather than per-vendor
+ SDKs. Formats supported for encoding are H.264, H.265 and VP9. Also
+ includes audio encoders for AAC and MP3. See Seungha’s blog post
+ “Bringing Microsoft Media Foundation to GStreamer” for some more
+ details about this.
+
+- new mfvideosrc video capture source element using the latest Windows
+ APIs rather than ancient APIs used by ksvideosrc/winks. ksvideosrc
+ should be considered deprecated going forward.
+
+- d3d11: add d3d11convert, a color space conversion and rescaling
+ element using shaders, and introduce d3d11upload and d3d11download
+ elements that work just like glupload and gldownload but for D3D11.
+
+- Universal Windows Platform (UWP) support, including official
+ GStreamer binary packages for it. Check out Nirbheek’s latest blog
+ post “GStreamer 1.18 supports the Universal Windows Platform” for
+ more details.
+
+- systemclock correctness and reliability fixes, and also don’t start
+ the system clock at 0 any longer (which shouldn’t make any
+ difference to anyone, as absolute clock time values are supposed to
+ be meaningless in themselves, only the rate of increase matters).
+
+- toolchain specific plugin registry: the registry cache is now named
+ differently for MSVC and MinGW toolchains/packages, which should
+ avoid problems when switching between binaries built with a
+ different toolchain.
+
+- new wasapi2 plugin mainly to support UWP applications. The core
+ logic of this plugin is almost identical to existing wasapi plugin,
+ but the main target is Windows 10 and UWP. This plugin uses WinRT
+ APIs, so will likely not work on Windows 8 or older. Unlike the
+ existing wasapi plugin, this plugin supports automatic stream
+ routing (auto fallback when device was removed) and device level
+ mute/volume control. Exclusive streaming mode is not supported,
+ however, and loopback features are not implemented yet. It is also
+ only possible to build this plugin with MSVC and the Windows 10 SDK,
+ it can’t be cross-compiled with the MingW toolchain.
+
+- new dxgiscreencapsrc element which uses the Desktop Duplication API
+ to capture the desktop screen at high speed. This is only supported
+ on Windows 8 or later. Compared to the existing elements
+ dxgiscreencapsrc offers much better performance, works in High DPI
+ environments and draws an accurate mouse cursor.
+
+- d3dvideosink was downgraded to secondary rank, d3d11videosink is
+ preferred now. Support OverlayComposition for GPU overlay
+ compositing of subtitles and logos.
+
+- debug log output fixes, esp. with a non-UTF8 locale/codepage
+
+- speex, jack: fixed crashes on Windows caused by cross-CRT issues
+
+- gst-play-1.0 interactive keyboard controls now also work on Windows
+
+Linux
+
+- kmssink: Add support for P010 and P016 formats
+
+- vah264dec: new experimental va plugin with an element for H.264
+ decoding with VA-API. This novel approach, different from
+ gstreamer-vaapi, uses the gstcodecs library for decoder state
+ handling, which it is hoped will make for cleaner code because it
+ uses VA-API without further layers or wrappers. Check out Víctor’s
+ blog post “New VA-API H.264 decoder in gst-plugins-bad” for the full
+ lowdown and the limitations of this new plugin, and how to give it a
+ spin.
+
+- v4l2codecs: introduce a V4L2 CODECs Accelerator. This plugin will
+ support the new CODECs uAPI in the Linux kernel, which consists of
+ an accelerator interface similar to DXVA, NVDEC, VDPAU and VAAPI. So
+ far H.264 and VP8 are supported. This is used on certain embedded
+ systems such as i.mx8m, rk3288, rk3399, Allwinner H-series SoCs.
+
+Documentation improvements
+
+- unified documentation containing tutorials, API docs, plugin docs,
+ etc. all under one roof, shipped in form of a documentation release
+ tarball containing both devhelp and html documentation.
+
+- all documentation is now generated using hotdoc, gtk-doc is no
+ longer used. Distributors should use the above-mentioned
+ documentation release tarball instead of trying to package hotdoc
+ and building the documentation from scratch.
+
+- there is now documentation for wrapper plugins like gst-libav and
+ frei0r, as well as tracer plugins.
+
+- for more info, check out Thibault’s “GStreamer Documentation”
+ lightning talk from the 2019 GStreamer Conference.
+
+- new API for plugins to support the documentation system:
+
+ - new GParamSpecFlag GST_PARAM_DOC_SHOW_DEFAULT to make
+ gst-inspect-1.0 (and the documentation) show the paramspec’s
+ default value rather than the actually set value as default
+ - GstPadTemplate getter and setter for “documentation caps”,
+ gst_pad_template_set_documentation_caps() and
+ gst_pad_template_get_documentation_caps(): This can be used in
+ elements where the caps of pad templates are dynamically
+ generated and/or dependent on the environment, to override the
+ caps shown in the documentation (usually to advertise the full
+ set of possible caps).
+ - gst_type_mark_as_plugin_api() for marking types as plugin API,
+ used for plugin-internal types like enums, flags, pad
+ subclasses, boxed types, and such.
+
+Possibly Breaking Changes
+
+- GstVideo: the canonical list of raw video formats (for use in caps)
+ has been reordered, so video elements such as videotestsrc or
+ videoconvert might negotiate to a different format now than before.
+ The new format might be a higher-quality format or require more
+ processing overhead, which might affect pipeline performance.
+
+- mpegtsdemux used to wrongly advertise H.264 and H.265 video
+ elementary streams as alignment=nal. This has now been fixed and
+ changed to alignment=none, which means an h264parse or h265parse
+ element is now required after tsdemux for some pipelines where there
+ wasn’t one before, e.g. in transmuxing scenarios (tsdemux ! tsmux).
+ Pipelines without such a parser may now fail to link or error out at
+ runtime. As parsers after demuxers and before muxers have been
+ generally required for a long time now it is hoped that this will
+ only affect a small number of applications or pipelines.
+
+- The Android opensles audio source and sink used to have hard-coded
+ buffer-/latency-time values of 20ms. This is no longer needed with
+ newer Android versions and has now been removed. This means a higher
+ or lower value might now be negotiated by default, which can affect
+ pipeline performance and latency.
Known Issues
-- The webrtcdsp element (which is unrelated to the newly-landed
- GStreamer webrtc support) is currently not shipped as part of the
- Windows binary packages due to a build system issue.
+- None in particular
+
+Contributors
+
+Aaron Boxer, Adam Duskett, Adam x Nilsson, Adrian Negreanu, Akinobu
+Mita, Alban Browaeys, Alcaro, Alexander Lapajne, Alexandru Băluț, Alex
+Ashley, Alex Hoenig, Alicia Boya García, Alistair Buxton, Ali Yousuf,
+Ambareesh “Amby” Balaji, Amr Mahdi, Andoni Morales Alastruey, Andreas
+Frisch, Andre Guedes, Andrew Branson, Andrey Sazonov, Antonio Ospite,
+aogun, Arun Raghavan, Askar Safin, AsociTon, A. Wilcox, Axel Mårtensson,
+Ayush Mittal, Bastian Bouchardon, Benjamin Otte, Bilal Elmoussaoui,
+Brady J. Garvin, Branko Subasic, Camilo Celis Guzman, Carlos Rafael
+Giani, Charlie Turner, Cheng-Chang Wu, Chris Ayoup, Chris Lord,
+Christoph Reiter, cketti, Damian Hobson-Garcia, Daniel Klamt, Daniel
+Molkentin, Danny Smith, David Bender, David Gunzinger, David Ing, David
+Svensson Fors, David Trussel, Debarshi Ray, Derek Lesho, Devarsh
+Thakkar, dhilshad, Dimitrios Katsaros, Dmitriy Purgin, Dmitry Shusharin,
+Dominique Leuenberger, Dong Il Park, Doug Nazar, dudengke, Dylan McCall,
+Dylan Yip, Ederson de Souza, Edward Hervey, Eero Nurkkala, Eike Hein,
+ekwange, Eric Marks, Fabian Greffrath, Fabian Orccon, Fabio D’Urso,
+Fabrice Bellet, Fabrice Fontaine, Fanchao L, Felix Yan, Fernando
+Herrrera, Francisco Javier Velázquez-García, Freyr, Fuwei Tang, Gaurav
+Kalra, George Kiagiadakis, Georgii Staroselskii, Georg Lippitsch, Georg
+Ottinger, gla, Göran Jönsson, Gordon Hart, Gregor Boirie, Guillaume
+Desmottes, Guillermo Rodríguez, Haakon Sporsheim, Haihao Xiang, Haihua
+Hu, Havard Graff, Håvard Graff, Heinrich Kruger, He Junyan, Henry
+Wilkes, Hosang Lee, Hou Qi, Hu Qian, Hyunjun Ko, ibauer, Ignacio Casal
+Quinteiro, Ilya Smelykh, Jake Barnes, Jakub Adam, James Cowgill, James
+Westman, Jan Alexander Steffens, Jan Schmidt, Jan Tojnar, Javier Celaya,
+Jeffy Chen, Jennifer Berringer, Jens Göpfert, Jérôme Laheurte, Jim
+Mason, Jimmy Ohn, J. Kim, Joakim Johansson, Jochen Henneberg, Johan
+Bjäreholt, Johan Sternerup, John Bassett, Jonas Holmberg, Jonas Larsson,
+Jonathan Matthew, Jordan Petridis, Jose Antonio Santos Cadenas, Josep
+Torra, Jose Quaresma, Josh Matthews, Joshua M. Doe, Juan Navarro,
+Juergen Werner, Julian Bouzas, Julien Isorce, Jun-ichi OKADA, Justin
+Chadwell, Justin Kim, Keri Henare, Kevin JOLY, Kevin King, Kevin Song,
+Knut Andre Tidemann, Kristofer Björkström, krivoguzovVlad, Kyrylo
+Polezhaiev, Lenny Jorissen, Linus Svensson, Loïc Le Page, Loïc Minier,
+Lucas Stach, Ludvig Rappe, Luka Blaskovic, luke.lin, Luke Yelavich,
+Marcin Kolny, Marc Leeman, Marco Felsch, Marcos Kintschner, Marek
+Olejnik, Mark Nauwelaerts, Markus Ebner, Martin Liska, Martin Theriault,
+Mart Raudsepp, Matej Knopp, Mathieu Duponchelle, Mats Lindestam, Matthew
+Read, Matthew Waters, Matus Gajdos, Maxim Paymushkin, Maxim P.
+Dementiev, Michael Bunk, Michael Gruner, Michael Olbrich, Miguel París
+Díaz, Mikhail Fludkov, Milian Wolff, Millan Castro, Muhammet Ilendemli,
+Nacho García, Nayana Topolsky, Nian Yan, Nicola Murino, Nicolas
+Dufresne, Nicolas Pernas Maradei, Niels De Graef, Nikita Bobkov, Niklas
+Hambüchen, Nirbheek Chauhan, Ognyan Tonchev, okuoku, Oleksandr
+Kvl,Olivier Crête, Ondřej Hruška, Pablo Marcos Oltra, Patricia Muscalu,
+Peter Seiderer, Peter Workman, Philippe Normand, Philippe Renon, Philipp
+Zabel, Pieter Willem Jordaan, Piotr Drąg, Ralf Sippl, Randy Li, Rasmus
+Thomsen, Ratchanan Srirattanamet, Raul Tambre, Ray Tiley, Richard
+Kreckel, Rico Tzschichholz, R Kh, Robert Rosengren, Robert Tiemann,
+Roman Shpuntov, Roman Sivriver, Ruben Gonzalez, Rubén Gonzalez,
+rubenrua, Ryan Huang, Sam Gigliotti, Santiago Carot-Nemesio, Saunier
+Thibault, Scott Kanowitz, Sebastian Dröge, Sebastiano Barrera, Seppo
+Yli-Olli, Sergey Nazaryev, Seungha Yang, Shinya Saito, Silvio
+Lazzeretti, Simon Arnling Bååth, Siwon Kang, sohwan.park, Song Bing,
+Soohyun Lee, Srimanta Panda, Stefano Buora, Stefan Sauer, Stéphane
+Cerveau, Stian Selnes, Sumaid Syed, Swayamjeet, Thiago Santos, Thibault
+Saunier, Thomas Bluemel, Thomas Coldrick, Thor Andreassen, Tim-Philipp
+Müller, Ting-Wei Lan, Tobias Ronge, trilene, Tulio Beloqui, U. Artie
+Eoff, VaL Doroshchuk, Varunkumar Allagadapa, Vedang Patel, Veerabadhran
+G, Víctor Manuel Jáquez Leal, Vivek R, Vivia Nikolaidou, Wangfei, Wang
+Zhanjun, Wim Taymans, Wonchul Lee, Xabier Rodriguez Calvar, Xavier
+Claessens, Xidorn Quan, Xu Guangxin, Yan Wang, Yatin Maan, Yeongjin
+Jeong, yychao, Zebediah Figura, Zeeshan Ali, Zeid Bekli, Zhiyuan Sraf,
+Zoltán Imets,
+
+… and many others who have contributed bug reports, translations, sent
+suggestions or helped testing.
+
+Stable 1.18 branch
+
+After the 1.18.0 release there will be several 1.18.x bug-fix releases
+which will contain bug fixes which have been deemed suitable for a
+stable branch, but no new features or intrusive changes will be added to
+a bug-fix release usually. The 1.18.x bug-fix releases will be made from
+the git 1.18 branch, which will be a stable branch.
+
+1.18.0
+1.18.0 was released on 7 September 2020.
-Schedule for 1.16
+Schedule for 1.20
-Our next major feature release will be 1.16, and 1.15 will be the
-unstable development version leading up to the stable 1.16 release. The
-development of 1.15/1.16 will happen in the git master branch.
+Our next major feature release will be 1.20, and 1.19 will be the
+unstable development version leading up to the stable 1.20 release. The
+development of 1.19/1.20 will happen in the git master branch.
-The plan for the 1.16 development cycle is yet to be confirmed, but it
-is expected that feature freeze will be around August 2018 followed by
-several 1.15 pre-releases and the new 1.16 stable release in September.
+The plan for the 1.20 development cycle is yet to be confirmed, but it
+is now expected that feature freeze will take place some time in January
+2021, with the first 1.20 stable release around February/March 2021.
-1.16 will be backwards-compatible to the stable 1.14, 1.12, 1.10, 1.8,
-1.6, 1.4, 1.2 and 1.0 release series.
+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.
------------------------------------------------------------------------
-_These release notes have been prepared by Tim-Philipp Müller with_
-_contributions from Sebastian Dröge, Sreerenj Balachandran, Thibault
-Saunier_ _and Víctor Manuel Jáquez Leal._
+These release notes have been prepared by Tim-Philipp Müller with
+contributions from Mathieu Duponchelle, Matthew Waters, Nirbheek
+Chauhan, Sebastian Dröge, Thibault Saunier, and Víctor Manuel Jáquez
+Leal.
-_License: CC BY-SA 4.0_
+License: CC BY-SA 4.0