-GSTREAMER 1.14 RELEASE NOTES
+GSTREAMER 1.16 RELEASE NOTES
-GStreamer 1.14.0 has not been released yet. It is scheduled for release
-in early March 2018.
+GStreamer 1.16.0 was originally released on 19 April 2019.
-There are unstable pre-releases available for testing and development
-purposes. The latest pre-release is version 1.13.91 (rc2) and was
-released on 12 March 2018.
-
-See https://gstreamer.freedesktop.org/releases/1.14/ for the latest
+See https://gstreamer.freedesktop.org/releases/1.16/ for the latest
version of this document.
-_Last updated: Monday 12 March 2018, 18:00 UTC (log)_
+_Last updated: Friday 19 April 2019, 00:00 UTC (log)_
Introduction
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.
Highlights
-- WebRTC support: real-time audio/video streaming to and from web
- browsers
+- GStreamer WebRTC stack gained support for data channels for
+ peer-to-peer communication based on SCTP, BUNDLE support, as well as
+ support for multiple TURN servers.
-- Experimental support for the next-gen royalty-free AV1 video codec
+- AV1 video codec support for Matroska and QuickTime/MP4 containers
+ and more configuration options and supported input formats for the
+ AOMedia AV1 encoder
-- Video4Linux: encoding support, stable element names and faster
- device probing
+- Support for Closed Captions and other Ancillary Data in video
-- Support for the Secure Reliable Transport (SRT) video streaming
- protocol
+- Support for planar (non-interleaved) raw audio
-- RTP Forward Error Correction (FEC) support (ULPFEC)
+- GstVideoAggregator, compositor and OpenGL mixer elements are now in
+ -base
-- RTSP 2.0 support in rtspsrc and gst-rtsp-server
+- New alternate fields interlace mode where each buffer carries a
+ single field
-- ONVIF audio backchannel support in gst-rtsp-server and rtspsrc
+- WebM and Matroska ContentEncryption support in the Matroska demuxer
-- playbin3 gapless playback and pre-buffering support
+- new WebKit WPE-based web browser source element
-- tee, our stream splitter/duplication element, now does allocation
- query aggregation which is important for efficient data handling and
- zero-copy
+- Video4Linux: HEVC encoding and decoding, JPEG encoding, and improved
+ dmabuf import/export
-- 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.
+- Hardware-accelerated Nvidia video decoder gained support for VP8/VP9
+ decoding, whilst the encoder gained support for H.265/HEVC encoding.
-- rtpjitterbuffer fast-start mode and timestamp offset adjustment
- smoothing
+- Many improvements to the Intel Media SDK based hardware-accelerated
+ video decoder and encoder plugin (msdk): dmabuf import/export for
+ zero-copy integration with other components; VP9 decoding; 10-bit
+ HEVC encoding; video post-processing (vpp) support including
+ deinterlacing; and the video decoder now handles dynamic resolution
+ changes.
-- souphttpsrc connection sharing, which allows for connection reuse,
- cookie sharing, etc.
+- The ASS/SSA subtitle overlay renderer can now handle multiple
+ subtitles that overlap in time and will show them on screen
+ simultaneously
-- nvdec: new plugin for hardware-accelerated video decoding using the
- NVIDIA NVDEC API
+- The Meson build is now feature-complete (*) and it is now the
+ recommended build system on all platforms. The Autotools build is
+ scheduled to be removed in the next cycle.
-- Adaptive DASH trick play support
+- The GStreamer Rust bindings and Rust plugins module are now
+ officially part of upstream GStreamer.
-- ipcpipeline: new plugin that allows splitting a pipeline across
- multiple processes
+- The GStreamer Editing Services gained a gesdemux element that allows
+ directly playing back serialized edit list with playbin or
+ (uri)decodebin
-- Major gobject-introspection annotation improvements for large parts
- of the library API
+- Many performance improvements
Major new features and changes
-WebRTC support
+Noteworthy new API
-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!
+- GstAggregator has a new "min-upstream-latency" property that forces
+ a minimum aggregate latency for the input branches of an aggregator.
+ This is useful for dynamic pipelines where branches with a higher
+ latency might be added later after the pipeline is already up and
+ running and where a change in the latency would be disruptive. This
+ only applies to the case where at least one of the input branches is
+ live though, it won’t force the aggregator into live mode in the
+ absence of any live inputs.
+
+- GstBaseSink gained a "processing-deadline" property and
+ setter/getter API to configure a processing deadline for live
+ pipelines. The processing deadline is the acceptable amount of time
+ to process the media in a live pipeline before it reaches the sink.
+ This is on top of the systemic latency that is normally reported by
+ the latency query. This defaults to 20ms and should make pipelines
+ such as v4l2src ! xvimagesink not claim that all frames are late in
+ the QoS events. Ideally, this should replace the "max-lateness"
+ property for most applications.
+
+- RTCP Extended Reports (XR) parsing according to RFC 3611:
+ Loss/Duplicate RLE, Packet Receipt Times, Receiver Reference Time,
+ Delay since the last Receiver (DLRR), Statistics Summary, and VoIP
+ Metrics reports. This only provides the ability to parse such
+ packets, generation of XR packets is not supported yet and XR
+ packets are not automatically parsed by rtpbin / rtpsession but must
+ be actively handled by the application.
+
+- a new mode for interlaced video was added where each buffer carries
+ a single field of interlaced video, with buffer flags indicating
+ whether the field is the top field or bottom field. Top and bottom
+ fields are expected to alternate in this mode. Caps for this
+ interlace mode must also carry a format:Interlaced caps feature to
+ ensure backwards compatibility.
+
+- The video library has gained support for three new raw pixel
+ formats:
-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.
+ - Y410: packed 4:4:4 YUV, 10 bits per channel
+ - Y210: packed 4:2:2 YUV, 10 bits per channel
+ - NV12_10LE40: fully-packed 10-bit variant of NV12_10LE32,
+ i.e. without the padding bits
+
+- GstRTPSourceMeta is a new meta that can be used to transport
+ information about the origin of depayloaded or decoded RTP buffers,
+ e.g. when mixing audio from multiple sources into a single stream. A
+ new "source-info" property on the RTP depayloader base class
+ determines whether depayloaders should put this meta on outgoing
+ buffers. Similarly, the same property on RTP payloaders determines
+ whether they should use the information from this meta to construct
+ the CSRCs list on outgoing RTP buffers.
-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.
+- gst_sdp_message_from_text() is a convenience constructor to parse
+ SDPs from a string which is particularly useful for language
+ bindings.
-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.
+Support for Planar (Non-Interleaved) Raw Audio
-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.
+Raw audio samples are usually passed around in interleaved form in
+GStreamer, which means that if there are multiple audio channels the
+samples for each channel are interleaved in memory, e.g.
+|LEFT|RIGHT|LEFT|RIGHT|LEFT|RIGHT| for stereo audio. A non-interleaved
+or planar arrangement in memory would look like
+|LEFT|LEFT|LEFT|RIGHT|RIGHT|RIGHT| instead, possibly with
+|LEFT|LEFT|LEFT| and |RIGHT|RIGHT|RIGHT| residing in separate memory
+chunks or separated by some padding.
-New Elements
+GStreamer has always had signalling for non-interleaved audio since
+version 1.0, but it was never actually properly implemented in any
+elements. audioconvert would advertise support for it, but wasn’t
+actually able to handle it correctly.
-- 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.
+With this release we now have full support for non-interleaved audio as
+well, which means more efficient integration with external APIs that
+handle audio this way, but also more efficient processing of certain
+operations like interleaving multiple 1-channel streams into a
+multi-channel stream which can be done without memory copies now.
-Noteworthy new API
+New API to support this has been added to the GStreamer Audio support
+library: There is now a new GstAudioMeta which describes how data is
+laid out inside the buffer, and buffers with non-interleaved audio must
+always carry this meta. To access the non-interleaved audio samples you
+must map such buffers with gst_audio_buffer_map() which works much like
+gst_buffer_map() or gst_video_frame_map() in that it will populate a
+little GstAudioBuffer helper structure passed to it with the number of
+samples, the number of planes and pointers to the start of each plane in
+memory. This function can also be used to map interleaved audio buffers
+in which case there will be only one plane of interleaved samples.
-- 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:
- Instant large changes to the internal ts_offset may cause timestamps
- to move backwards and 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
- continue streaming if one of the inputs stops producing data.
-
-- jpegenc has gained a "snapshot" property just like pngenc to make it
- easier to just output a single encoded frame.
-
-- jpegdec will now handle interlaced MJPEG streams properly and also
- handle 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 but adding new codecs is just a
- matter of defining appropriate maximum frame sizes.
-
-- 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, 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 had 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. More tuning options 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
+Of course support for this has also been implemented in the various
+audio helper and conversion APIs, base classes, and in elements such as
+audioconvert, audioresample, audiotestsrc, audiorate.
-Plugin and library moves
+Support for Closed Captions and Other Ancillary Data in Video
+
+The video support library has gained support for detecting and
+extracting Ancillary Data from videos as per the SMPTE S291M
+specification, including:
+
+- a VBI (Vertical Blanking Interval) parser that can detect and
+ extract Ancillary Data from Vertical Blanking Interval lines of
+ component signals. This is currently supported for videos in v210
+ and UYVY format.
+
+- a new GstMeta for closed captions: GstVideoCaptionMeta. This
+ supports the two types of closed captions, CEA-608 and CEA-708,
+ along with the four different ways they can be transported (other
+ systems are a superset of those).
+
+- a VBI (Vertical Blanking Interval) encoder for writing ancillary
+ data to the Vertical Blanking Interval lines of component signals.
+
+The new closedcaption plugin in gst-plugins-bad then makes use of all
+this new infrastructure and provides the following elements:
+
+- cccombiner: a closed caption combiner that takes a closed captions
+ stream and another stream and adds the closed captions as
+ GstVideoCaptionMeta to the buffers of the other stream.
+
+- ccextractor: a closed caption extractor which will take
+ GstVideoCaptionMeta from input buffers and output them as a separate
+ closed captions stream.
+
+- ccconverter: a closed caption converter that can convert between
+ different formats
+
+- line21encoder, line21decoder: inject/extract line21 closed captions
+ to/from SD video streams
+
+- cc708overlay: decodes CEA 608/708 captions and overlays them on
+ video
+
+Additionally, the following elements have also gained Closed Caption
+support:
+
+- qtdemux and qtmux support CEA 608/708 Closed Caption tracks
+
+- mpegvideoparse, h264parse extracts Closed Captions from MPEG-2/H.264
+ video streams
+
+- avviddec, avvidenc, x264enc got support for extracting/injecting
+ Closed Captions
+
+- decklinkvideosink can output closed captions and decklinkvideosrc
+ can extract closed captions
+
+- playbin and playbin3 learned how to autoplug CEA 608/708 CC overlay
+ elements
-MPEG-1 audio (mp1, mp2, mp3) decoders and encoders moved to -good
+- the externally maintained ajavideosrc element for AJA capture cards
+ has support for extracting closed captions
-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.
+The rsclosedcaption plugin in the Rust plugins collection includes a
+MacCaption (MCC) file parser and encoder.
-In particular these are:
+New Elements
+
+- overlaycomposition: New element that allows applications to draw
+ GstVideoOverlayCompositions on a stream. The element will emit the
+ "draw" signal for each video buffer, and the application then
+ generates an overlay for that frame (or not). This is much more
+ performant than e.g. cairooverlay for many use cases, e.g. because
+ pixel format conversions can be avoided or the blitting of the
+ overlay can be delegated to downstream elements (such as
+ gloverlaycompositor). It’s particularly useful for cases where only
+ a small section of the video frame should be drawn on.
+
+- gloverlaycompositor: New OpenGL-based compositor element that
+ flattens any overlays from GstVideoOverlayCompositionMetas into the
+ video stream. This element is also always part of glimagesink.
+
+- glalpha: New element that adds an alpha channel to a video stream.
+ The values of the alpha channel can either be set to a constant or
+ can be dynamically calculated via chroma keying. It is similar to
+ the existing alpha element but based on OpenGL. Calculations are
+ done in floating point so results may not be identical to the output
+ of the existing alpha element.
+
+- rtpfunnel funnels together RTP streams into a single session. Use
+ cases include multiplexing and bundle. webrtcbin uses it to
+ implement BUNDLE support.
+
+- testsrcbin is a source element that provides an audio and/or video
+ stream and also announces them using the recently-introduced
+ GstStream API. This is useful for testing elements such as playbin3
+ or uridecodebin3 etc.
+
+- New closed caption elements: cccombiner, ccextractor, ccconverter,
+ line21encoder, line21decoder and cc708overlay (see above)
+
+- wpesrc: new source element acting as a Web Browser based on WebKit
+ WPE
+
+- Two new OpenCV-based elements: cameracalibrate and cameraundistort
+ that can communicate to figure out distortion correction parameters
+ for a camera and correct for the distortion.
+
+- New sctp plugin based on usrsctp with sctpenc and sctpdec elements.
+ These elements are used inside webrtcbin for implementing data
+ channels.
+
+New element features and additions
+
+- playbin3, playbin and playsink have gained a new "text-offset"
+ property to adjust the positioning of the selected subtitle stream
+ vis-a-vis the audio and video streams. This uses subtitleoverlay’s
+ new "subtitle-ts-offset" property. GstPlayer has gained matching API
+ for this, namely gst_player_get_text_video_offset().
+
+- playbin3 buffering improvements: in network playback scenarios there
+ may be multiple inputs to decodebin3, and buffering will be done
+ before decodebin3 using queue2 or downloadbuffer elements inside
+ urisourcebin. Since this is before any parsers or demuxers there may
+ not be any bitrate information available for the various streams, so
+ it was difficult to configure the buffering there smartly within
+ global constraints. This was improved now: The queue2 elements
+ inside urisourcebin will now use the new bitrate query to figure out
+ a bitrate estimate for the stream if no bitrate was provided by
+ upstream, and urisourcebin will use the bitrates of the individual
+ queues to distribute the globally-set "buffer-size" budget in bytes
+ to the various queues. urisourcebin also gained "low-watermark" and
+ "high-watermark" properties which will be proxied to the internal
+ queues, as well as a read-only "statistics" property which allows
+ querying of the minimum/maximum/average byte and time levels of the
+ queues inside the urisourcebin in question.
+
+- splitmuxsink has gained a couple of new features:
+
+ - new "async-finalize" mode: This mode is useful for muxers or
+ outputs that can take a long time to finalize a file. Instead of
+ blocking the whole upstream pipeline while the muxer is doing
+ its stuff, we can unlink it and spawn a new muxer + sink
+ combination to continue running normally. This requires us to
+ receive the muxer and sink (if needed) as factories via the new
+ "muxer-factory" and "sink-factory" properties, optionally
+ accompanied by their respective properties structures (set via
+ the new "muxer-properties" and "sink-properties" properties).
+ There are also new "muxer-added" and "sink-added" signals in
+ case custom code has to be called for them to configure them.
+
+ - "split-at-running-time" action signal: When called by the user,
+ this action signal ends the current file (and starts a new one)
+ as soon as the given running time is reached. If called multiple
+ times, running times are queued up and processed in the order
+ they were given.
+
+ - "split-after" action signal to finish outputting the current GOP
+ to the current file and then start a new file as soon as the GOP
+ is finished and a new GOP is opened (unlike the existing
+ "split-now" which immediately finishes the current file and
+ writes the current GOP into the next newly-started file).
+
+ - "reset-muxer" property: when unset, the muxer is reset using
+ flush events instead of setting its state to NULL and back. This
+ means the muxer can keep state across resets, e.g. mpegtsmux
+ will keep the continuity counter continuous across segments as
+ required by hlssink2.
+
+- qtdemux gained PIFF track encryption box support in addition to the
+ already-existing PIFF sample encryption support, and also allows
+ applications to select which encryption system to use via a
+ "drm-preferred-decryption-system-id" context in case there are
+ multiple options.
+
+- qtmux: the "start-gap-threshold" property determines now whether an
+ edit list will be created to account for small gaps or offsets at
+ the beginning of a stream in case the start timestamps of tracks
+ don’t line up perfectly. Previously the threshold was hard-coded to
+ 1% of the (video) frame duration, now it is 0 by default (so edit
+ list will be created even for small differences), but fully
+ configurable.
+
+- rtpjitterbuffer has improved end-of-stream handling
+
+- rtpmp4vpay will be prefered over rtpmp4gpay for MPEG-4 video in
+ autoplugging scenarios now
+
+- rtspsrc now allows applications to send RTSP SET_PARAMETER and
+ GET_PARAMETER requests using action signals.
+
+- rtspsrc has a small (100ms) configurable teardown delay by default
+ to try and make sure an RTSP TEARDOWN request gets sent out when the
+ source element shuts down. This will block the downward PAUSED to
+ READY state change for a short time, but can be disabled where it’s
+ a problem. Some servers only allow a limited number of concurrent
+ clients, so if no proper TEARDOWN is sent new clients may have
+ problems connecting to the server for a while.
+
+- souphttpsrc behaves better with low bitrate streams now. Before it
+ would increase the read block size too quickly which could lead to
+ it not reading any data from the socket for a very long time with
+ low bitrate streams that are output live downstream. This could lead
+ to servers kicking off the client.
+
+- filesink: do internal buffering to avoid performance regression with
+ small writes since we bypass libc buffering by using writev()
+ instead of fwrite()
+
+- identity: add "eos-after" property and fix "error-after" property
+ when the element is reused
+
+- input-selector: lets context queries pass through, so that
+ e.g. upstream OpenGL elements can use contexts and displays
+ advertised by downstream elements
+
+- queue2: avoid ping-pong between 0% and 100% buffering messages if
+ upstream is pushing buffers larger than one of its limits, plus
+ performance optimisations
+
+- opusdec: new "phase-inversion" property to control phase inversion.
+ When enabled, this will slightly increase stereo quality, but
+ produces a stream that when downmixed to mono will suffer audio
+ distortions.
+
+- The x265enc HEVC encoder also exposes a "key-int-max" property to
+ configure the maximum allowed GOP size now.
+
+- decklinkvideosink has seen stability improvements for long-running
+ pipelines (potential crash due to overflow of leaked clock refcount)
+ and clock-slaving improvements when performing flushing seeks
+ (causing stalls in the output timeline), pausing and/or buffering.
+
+- srtpdec, srtpenc: add support for MKIs which allow multiple keys to
+ be used with a single SRTP stream
+
+- srtpdec, srtpenc: add support for AES-GCM and also add support for
+ it in gst-rtsp-server and rtspsrc.
+
+- The srt Secure Reliable Transport plugin has integrated server and
+ client elements srt{client,server}{src,sink} into one (srtsrc and
+ srtsink), since SRT connection mode can be changed by uri
+ parameters.
+
+- h264parse and h265parse will handle SEI recovery point messages and
+ mark recovery points as keyframes as well (in addition to IDR
+ frames)
+
+- webrtcbin: "add-turn-server" action signal to pass multiple ICE
+ relays (TURN servers).
+
+- The removesilence element has received various new features and
+ properties, such as a "threshold" property, detecting silence only
+ after minimum silence time/buffers, a "silent" property to control
+ bus message notifications as well as a "squash" property.
+
+- AOMedia AV1 decoder gained support for 10/12bit decoding whilst the
+ AV1 encoder supports more image formats and subsamplings now and
+ acquired support for rate control and profile related configuration.
+
+- The Fraunhofer fdkaac plugin can now be built against the 2.0.0
+ version API and has improved multichannel support
+
+- kmssink now supports unpadded 24-bit RGB and can configure mode
+ setting from video info, which enables display of multi-planar
+ formats such as I420 or NV12 with modesetting. It has also gained a
+ number of new properties: The "restore-crtc" property does what it
+ says on the tin and is enabled by default. "plane-properties" and
+ "connector-properties" can be used to pass custom properties to the
+ DRM.
+
+- waylandsink has a "fullscreen" property now and supports the
+ XDG-Shell protocol.
+
+- decklinkvideosink, decklinkvideosrc support selecting between
+ half/full duplex
-- mpg123audiodec: an mp1/mp2/mp3 audio decoder using libmpg123
-- lamemp3enc: an mp3 encoder using LAME
-- twolamemp2enc: an mp2 encoder using TwoLAME
+- The vulkan plugin gained support for macOS and iOS via MoltenVK in
+ addition to the existing support for X11 and Wayland
-GstAggregator moved from -bad to core
+- imagefreeze has a new num-buffers property to limit the number of
+ buffers that are produced and to send an EOS event afterwards
-GstAggregator has been moved from gst-plugins-bad to the base library in
-GStreamer and is now stable API.
+- webrtcbin has a new, introspectable get-transceiver signal in
+ addition to the old get-transceivers signal that couldn’t be used
+ from bindings
+
+- Support for per-element latency information was added to the latency
+ tracer
+
+Plugin and library moves
-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.
+- The stereo element was moved from -bad into the existing audiofx
+ plugin in -good. If you get duplicate type registration warnings
+ when upgrading, check that you don’t have a stale stereoplugin lying
+ about somewhere.
-GstAudioAggregator, audiomixer and audiointerleave moved from -bad to -base
+GstVideoAggregator, compositor, and OpenGL mixer elements 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.
+GstVideoAggregator is a new base class for raw video mixers and muxers
+and is based on GstAggregator. It provides defined-latency mixing of raw
+video 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 fixed; 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.
+Most notably, the "ignore-eos" pad property was renamed to
+"repeat-after-eos" and the conversion code was moved to a
+GstVideoAggregatorConvertPad subclass to avoid code duplication, make
+things less awkward for subclasses like the OpenGL-based video mixer,
+and make the API more consistent with the audio aggregator API.
+
+It is used by the compositor element, which is a replacement for
+‘videomixer’ which did not handle live inputs very well. compositor
+should behave much better in that respect and generally behave as one
+would expected in most scenarios.
+
+The compositor element has gained support for per-pad blending mode
+operators (SOURCE, OVER, ADD) which determines what operator to use for
+blending this pad over the previous ones. This can be used to implement
+crossfading and the available operators can be extended in the future as
+needed.
+
+A number of OpenGL-based video mixer elements (glvideomixer, glmixerbin,
+glvideomixerelement, glstereomix, glmosaic) which are built on top of
+GstVideoAggregator have also been moved from -bad to -base now. These
+elements have been merged into the existing OpenGL plugin, so if you get
+duplicate type registration warnings when upgrading, check that you
+don’t have a stale openglmixers plugin lying about somewhere.
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 plugins have been removed from gst-plugins-bad:
+
+- The experimental daala plugin has been removed, since it’s not so
+ useful now that all effort is focused on AV1 instead, and it had to
+ be enabled explicitly with --enable-experimental anyway.
+
+- The spc plugin has been removed. It has been replaced by the gme
+ plugin.
+
+- The acmmp3dec and acmenc plugins for Windows have been removed. ACM
+ is an ancient legacy API and there was no point in keeping the
+ plugins around for a licensed MP3 decoder now that the MP3 patents
+ have expired and we have a decoder in -good. We also didn’t ship
+ these in our cerbero-built Windows packages, so it’s unlikely that
+ they’ll be missed.
+
+
+Miscellaneous API additions
+
+- GstBitwriter: new generic bit writer API to complement the existing
+ bit reader
+
+- gst_buffer_new_wrapped_bytes() creates a wrap buffer from a GBytes
+
+- gst_caps_set_features_simple() sets a caps feature on all the
+ structures of a GstCaps
+
+- New GST_QUERY_BITRATE query: This allows determining from downstream
+ what the expected bitrate of a stream may be which is useful in
+ queue2 for setting time based limits when upstream does not provide
+ timing information. tsdemux, qtdemux and matroskademux have basic
+ support for this query on their sink pads.
+
+- elements: there is a new “Hardware” class specifier. Elements
+ interacting with hardware devices should specify this classifier in
+ their element factory class metadata. This is useful to advertise as
+ one might need to put such elements into READY state to test if the
+ hardware is present in the system for example.
+
+- protection: Add a new definition for unspecified system protection,
+ GST_PROTECTION_UNSPECIFIED_SYSTEM_ID
+
+- take functions for various mini objects that didn’t have them yet:
+ gst_query_take(), gst_message_take(), gst_tag_list_take(),
+ gst_buffer_list_take(). Unlike the various _replace() functions
+ _take() does not increase the reference count but takes ownership of
+ the mini object passed.
+
+- clear functions for various mini object types and GstObject which
+ unrefs the object or mini object (if non-NULL) and sets the variable
+ pointed to to NULL: gst_clear_structure(), gst_clear_tag_list(),
+ gst_clear_query(), gst_clear_message(), gst_clear_event(),
+ gst_clear_caps(), gst_clear_buffer_list(), gst_clear_buffer(),
+ gst_clear_mini_object(), gst_clear_object()
+
+- miniobject: new API gst_mini_object_add_parent() and
+ gst_mini_object_remove_parent() to set parent pointers on mini
+ objects to ensure correct writability: Every container of
+ miniobjects now needs to store itself as parent in the child object,
+ and remove itself again later. A mini object is then only writable
+ if there is at most one parent, that parent is writable itself, and
+ the reference count of the mini object is 1. GstBuffer (for
+ memories), GstBufferList (for buffers), GstSample (for caps, buffer,
+ bufferlist), and GstVideoOverlayComposition were updated
+ accordingly. Without this it was possible to have e.g. a buffer list
+ with a refcount of 2 used in two places at once that both modify the
+ same buffer with refcount 1 at the same time wrongly thinking it is
+ writable even though it’s really not.
+
+- poll: add API to watch for POLLPRI and stop treating POLLPRI as a
+ read. This is useful to wait for video4linux events which are
+ signalled via POLLPRI.
+
+- sample: new API to update the contents of a GstSample and make it
+ writable: gst_sample_set_buffer(), gst_sample_set_caps(),
+ gst_sample_set_segment(), gst_sample_set_info(), plus
+ gst_sample_is_writable() and gst_sample_make_writable(). This makes
+ it possible to reuse a sample object and avoid unnecessary memory
+ allocations, for example in appsink.
+
+- ClockIDs now keep a weak reference to underlying clock to avoid
+ crashes in basesink in corner cases where a clock goes away while
+ the ClockID is still in use, plus some new API
+ (gst_clock_id_get_clock(), gst_clock_id_uses_clock()) to check the
+ clock a ClockID is linked to.
+
+- The GstCheck unit test library gained a
+ fail_unless_equals_clocktime() convenience macro as well as some new
+ GstHarness API for for proposing meta APIs from the allocation
+ query: gst_harness_add_propose_allocation_meta(). ASSERT_CRITICAL()
+ checks in unit tests are now skipped if GStreamer was compiled with
+ GST_DISABLE_GLIB_CHECKS.
+
+- gst_audio_buffer_truncate() convenience function to truncate a raw
+ audio buffer
+
+- GstDiscoverer has support for caching the results of discovery in
+ the default cache directory. This can be enabled with the use-cache
+ property and is disabled by default.
+
+- GstMeta that are attached to GstBuffers are now always stored in the
+ order in which they were added.
+
+- Additional support for signalling ONVIF specific features were
+ added: the SEEK event can store a trickmode-interval now and support
+ for the Rate-Control and Frames RTSP headers was added to the RTSP
+ library.
+
+
+Miscellaneous performance and memory optimisations
+
+As always there have been many performance and memory usage improvements
+across all components and modules. Some of them (such as dmabuf
+import/export) have already been mentioned elsewhere so won’t be
+repeated here.
-- 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.
+The following list is only a small snapshot of some of the more
+interesting optimisations that haven’t been mentioned in other contexts
+yet:
-API removals
+- The GstVideoEncoder and GstVideoDecoder base classes now release the
+ STREAM_LOCK when pushing out buffers, which means (multi-threaded)
+ encoders and decoders can now receive and continue to process input
+ buffers whilst waiting for downstream elements in the pipeline to
+ process the buffer that was pushed out. This increases throughput
+ and reduces processing latency, also and especially for
+ hardware-accelerated encoder/decoder elements.
-- some MPEG video parser API in the API unstable codecutils library in
- gst-plugins-bad was removed after having been deprecated for 5
- years.
+- GstQueueArray has seen a few API additions
+ (gst_queue_array_peek_nth(), gst_queue_array_set_clear_func(),
+ gst_queue_array_clear()) so that it can be used in other places like
+ GstAdapter instead of a GList, which reduces allocations and
+ improves performance.
+
+- appsink now reuses the sample object in pull_sample() if possible
+
+- rtpsession only starts the RTCP thread when it’s actually needed now
+
+- udpsrc uses a buffer pool now and the GstUdpSrc object structure was
+ optimised for better cache performance
+
+GstPlayer
+
+- API was added to fine-tune the synchronisation offset between
+ subtitles and video
Miscellaneous changes
-- 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)
-
-- 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).
-
-- 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.
-
-- 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.
+- As a result of moving to newer FFmpeg APIs, encoder and decoder
+ elements exposed by the GStreamer FFmpeg wrapper plugin (gst-libav)
+ may have seen possibly incompatible changes to property names and/or
+ types, and not all properties exposed might be functional. We are
+ still reviewing the new properties and aim to minimise breaking
+ changes at least for the most commonly-used properties, so please
+ report any issues you run into!
OpenGL integration
-- The GStreamer OpenGL integration library has moved to
- gst-plugins-base and is now part of our stable API.
+- The OpenGL mixer elements have been moved from -bad to
+ gst-plugins-base (see above)
+
+- The Mesa GBM backend now supports headless mode
+
+- gloverlaycompositor: New OpenGL-based compositor element that
+ flattens any overlays from GstVideoOverlayCompositionMetas into the
+ video stream.
-- 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.
+- glalpha: New element that adds an alpha channel to a video stream.
+ The values of the alpha channel can either be set to a constant or
+ can be dynamically calculated via chroma keying. It is similar to
+ the existing alpha element but based on OpenGL. Calculations are
+ done in floating point so results may not be identical to the output
+ of the existing alpha element.
-- Prefer wayland display over X11: As most Wayland compositors support
- XWayland, the X11 backend would get selected.
+- glupload: Implement direct dmabuf uploader, the idea being that some
+ GPUs (like the Vivante series) can actually perform the YUV->RGB
+ conversion internally, so no custom conversion shaders are needed.
+ To make use of this feature, we need an additional uploader that can
+ import DMABUF FDs and also directly pass the pixel format, relying
+ on the GPU to do the conversion.
-- gldownload can export dmabufs now, and glupload will advertise
- dmabuf as caps feature.
+- The OpenGL library no longer restores the OpenGL viewport. This is a
+ performance optimization to not require performing multiple
+ expensive glGet*() function calls per frame. This affects any
+ application or plugin use of the following functions and objects:
+ - glcolorconvert library object (not the element)
+ - glviewconvert library object (not the element)
+ - gst_gl_framebuffer_draw_to_texture()
+ - custom GstGLWindow implementations
Tracing framework and debugging improvements
-- 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.
-
-- 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 use 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.
+- There is now a GDB PRETTY PRINTER FOR VARIOUS GSTREAMER TYPES: For
+ GstObject pointers the type and name is added, e.g.
+ 0x5555557e4110 [GstDecodeBin|decodebin0]. For GstMiniObject pointers
+ the object type is added, e.g. 0x7fffe001fc50 [GstBuffer]. For
+ GstClockTime and GstClockTimeDiff the time is also printed in human
+ readable form, e.g. 150116219955 [+0:02:30.116219955].
+- GDB EXTENSION WITH TWO CUSTOM GDB COMMANDS gst-dot AND gst-print:
-Tools
+ - gst-dot creates dot files that a very close to what
+ GST_DEBUG_BIN_TO_DOT_FILE() produces, but object properties and
+ buffer contents such as codec-data in caps are not available.
+
+ - gst-print produces high-level information about a GStreamer
+ object. This is currently limited to pads for GstElements and
+ events for the pads. The output may look like this:
-- 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_structure_to_string() now serialises the actual value of
+ pointers when serialising GstStructures instead of claiming they’re
+ NULL. This makes debug logging in various places less confusing,
+ because it’s clear now that structure fields actually hold valid
+ objects. Such object pointer values will never be deserialised
+ however.
-- gst-launch-1.0 now generates a gstreamer pipeline diagram (.dot
- file) whenever SIGHUP is sent to it on Linux/*nix systems.
-- gst-discoverer-1.0 can now analyse live streams such as rtsp:// URIs
+Tools
+
+- gst-inspect-1.0 has coloured output now and will automatically use a
+ pager if the output does not fit on a page. This only works in a
+ UNIX environment and if the output is not piped, and on Windows 10
+ build 16257 or newer. If you don’t like the colours you can disable
+ them by setting the GST_INSPECT_NO_COLORS=1 environment variable or
+ passing the --no-color command line option.
GStreamer RTSP server
-- Initial support for [RTSP protocol version
- 2.0][rtsp2-lightning-talk] was added, which is to the best of our
- knowledge the first RTSP 2.0 implementation ever!
-
-- 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.
-
-
-- 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.
-
-- rtspclientsink has gained an "accept-certificate" signal for
- manually checking a TLS certificate for validity.
-
-- 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.
-
-- 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.
-
-- Fix corner case issue where the wrong mount point was found when
- there were multiple mount points with a common prefix.
+- Improved backlog handling when using TCP interleaved for data
+ transport. Before there was a fixed maximum size for backlog
+ messages, which was prone to deadlocks and made it difficult to
+ control memory usage with the watch backlog. The RTSP server now
+ limits queued TCP data messages to one per stream, moving queuing of
+ the data into the pipeline and leaving the RTSP connection
+ responsive to RTSP messages in both directions, preventing all those
+ problems.
+
+- Initial ULP Forward Error Correction support in rtspclientsink and
+ for RECORD mode in the server.
+
+- API to explicitly enable retransmission requests (RTX)
+
+- Lots of multicast-related fixes
+
+- rtsp-auth: Add support for parsing .htdigest files
GStreamer VAAPI
-- this section will be filled in shortly {FIXME!}
+- Support Wayland’s display for context sharing, so the application
+ can pass its own wl_display in order to be used for the VAAPI
+ display creation.
+
+- A lot of work to support new Intel hardware using media-driver as VA
+ backend.
+
+- For non-x86 devices, VAAPI display can instantiate, through DRM,
+ with no PCI bus. This enables the usage of libva-v4l2-request
+ driver.
+
+- Added support for XDG-shell protocol as wl_shell replacement which
+ is currently deprecated. This change add as dependency
+ wayland-protocol.
+
+- GstVaapiFilter, GstVaapiWindow, and GstVaapiDecoder classes now
+ inherit from GstObject, gaining all the GStreamer’s instrumentation
+ support.
+
+- The metadata now specifies the plugin as Hardware class.
+
+- H264 decoder is more stable with problematic streams.
+
+- In H265 decoder added support for profiles main-422-10 (P010_10LE),
+ main-444 (AYUV) and main-444-10 (Y410)
+
+- JPEG decoder handles dynamic resolution changes.
+
+- More specification adherence in H264 and H265 encoders.
+
+
+GStreamer OMX
+
+- Add support of NV16 format to video encoders input.
+
+- Video decoders now handle the ALLOCATION query to tell upstream
+ about the number of buffers they require. Video encoders will also
+ use this query to adjust their number of allocated buffers
+ preventing starvation when using dynamic buffer mode.
+
+- The OMX_PERFORMANCE debug category has been renamed to OMX_API_TRACE
+ and can now be used to track a widder variety of interactions
+ between OMX and GStreamer.
+
+- Video encoders will now detect frame rate only changes and will
+ inform OMX about it rather than doing a full format reset.
+
+- Various Zynq UltraScale+ specific improvements:
+ - Video encoders are now able to import dmabuf from upstream.
+ - Support for HEVC range extension profiles and more AVC profiles.
+ - We can now request video encoders to generate an IDR using the
+ force key unit event.
GStreamer Editing Services and NLE
-- this section will be filled in shortly {FIXME!}
+- Added a gesdemux element, it is an auto pluggable element that
+ allows decoding edit list like files supported by GES
+
+- Added gessrc which wraps a GESTimeline as a standard source element
+ (implementing the ges protocol handler)
+
+- Added basic support for videorate::rate property potentially
+ allowing changing playback speed
+
+- Layer priority is now fully automatic and they should be moved with
+ the new ges_timeline_move_layer method, ges_layer_set_priority is
+ now deprecated.
+
+- Added a ges_timeline_element_get_layer_priority so we can simply get
+ all information about GESTimelineElement position in the timeline
+
+- GESVideoSource now auto orientates the images if it is defined in a
+ meta (overridable).
+
+- Added some PyGObject overrides to make the API more pythonic
+
+- The threading model has been made more explicit with safe guard to
+ make sure not thread safe APIs are not used from the wrong threads.
+ It is also now possible to properly handle in what thread the API
+ should be used.
+
+- Optimized GESClip and GESTrackElement creation
+
+- Added a way to compile out the old, unused and deprecated
+ GESPitiviFormatter
+
+- Re implemented the timeline editing API making it faster and making
+ the code much more maintainable
+
+- Simplified usage of nlecomposition outside GES by removing quirks in
+ it API usage and removing the need to treat it specially from an
+ application perspective.
+
+- ges-launch-1.0:
+
+ - Added support to add titles to the timeline
+ - Enhance the help auto generating it from the code
+
+- Deprecate ges_timeline_load_from_uri as loading the timeline should
+ be done through a project now
+
+- MANY leaks have been plugged and the unit testsuite is now “leak
+ free”
GStreamer validate
-- this section will be filled in shortly {FIXME!}
+- Added an action type to verify the checksum of the sink last-sample
+
+- Added an include keyword to validate scenarios
+
+- Added the notion of variable in scenarios, with the set-vars keyword
+
+- Started adding support for “performance” like tests by allowing to
+ define the number of dropped buffers or the minimum buffer frequency
+ on a specific pad
+
+- Added a validateflow plugin which allows defining the data flow to
+ be seen on a particular pad and verifying that following runs match
+ the expectations
+
+- Added support for appsrc based test definition so we can instrument
+ the data pushed into the pipeline from scenarios
+
+- Added a mockdecryptor allowing adding tests with on encrypted files,
+ the element will potentially be instrumented with a validate
+ scenario
+
+- gst-validate-launcher:
+
+ - Cleaned up output
+
+ - Changed the default for “muting” tests as user doesn’t expect
+ hundreds of windows to show up when running the testsuite
+
+ - Fixed the outputted xunit files to be compatible with GitLab
+
+ - Added support to run tests on media files in push mode (using
+ pushfile://)
+
+ - Added support for running inside gst-build
+
+ - Added support for running ssim tests on rendered files
+
+ - Added a way to simply define tests on pipelines through a simple
+ .json file
+
+ - Added a python app to easily run python testsuite reusing all
+ the launcher features
+
+ - Added flatpak knowledge so we can print backtrace even when
+ running from within flatpak
+
+ - Added a way to automatically generated “known issues”
+ suppressions lines
+
+ - Added a way to rerun tests to check if they are flaky and added
+ a way to tolerate tests known to be flaky
+
+ - Add a way to output html log files
GStreamer Python Bindings
-- this section will be filled in shortly {FIXME!}
+- add binding for gst_pad_set_caps()
+- pygobject dependency requirement was bumped to >= 3.8
-Build and Dependencies
+- new audiotestsrc, audioplot, and mixer plugin examples, and a
+ dynamic pipeline example
-- 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.
-- gst-plugins-bad no longer depends on the libschroedinger Dirac codec
- library.
+GStreamer C# Bindings
-- The srtp plugin can now also be built against libsrtp2.
-
-- 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.
-
-- 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 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.
-
-
-Note to packagers and distributors
-
-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.
-
-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.
-
-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).
-
-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.
-
-
-Platform-specific improvements
+- bindings for the GstWebRTC library
-Android
-- ahcsrc (Android camera source) does autofocus now
+GStreamer Rust Bindings
-macOS and iOS
+The GStreamer Rust bindings are now officially part of the GStreamer
+project and are also maintained in the GStreamer GitLab.
-- this section will be filled in shortly {FIXME!}
+The releases will generally not be synchronized with the releases of
+other GStreamer parts due to dependencies on other projects.
-Windows
+Also unlike the other GStreamer libraries, the bindings will not commit
+to full API stability but instead will follow the approach that is
+generally taken by Rust projects, e.g.:
-- 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:
+1) 0.12.X will be completely API compatible with all other 0.12.Y
+ versions.
+2) 0.12.X+1 will contain bugfixes and compatible new feature additions.
+3) 0.13.0 will _not_ be backwards compatible with 0.12.X but projects
+ will be able to stay at 0.12.X without any problems as long as they
+ don’t need newer features.
-- support for more than 2 channels
+The current stable release is 0.12.2 and the next release series will be
+0.13, probably around March 2019.
-- a new "low-latency" property to enable low-latency operation (which
- should always be safe to enable)
+At this point the bindings cover most of GStreamer core (except for most
+notably GstAllocator and GstMemory), and most parts of the app, audio,
+base, check, editing-services, gl, net. pbutils, player, rtsp,
+rtsp-server, sdp, video and webrtc libraries.
-- 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.
+Also included is support for creating subclasses of the following types
+and writing GStreamer plugins:
-- set realtime thread priority to avoid glitches
+- gst::Element
+- gst::Bin and gst::Pipeline
+- gst::URIHandler and gst::ChildProxy
+- gst::Pad, gst::GhostPad
+- gst_base::Aggregator and gst_base::AggregatorPad
+- gst_base::BaseSrc and gst_base::BaseSink
+- gst_base::BaseTransform
-- 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.
+Changes to 0.12.X since 0.12.0
-- 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
+Fixed
-- 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.
+- PTP clock constructor actually creates a PTP instead of NTP clock
+Added
-Contributors
+- Bindings for GStreamer Editing Services
+- Bindings for GStreamer Check testing library
+- Bindings for the encoding profile API (encodebin)
-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.
+- VideoFrame, VideoInfo, AudioInfo, StructureRef implements Send and
+ Sync now
+- VideoFrame has a function to get the raw FFI pointer
+- From impls from the Error/Success enums to the combined enums like
+ FlowReturn
+- Bin-to-dot file functions were added to the Bin trait
+- gst_base::Adapter implements SendUnique now
+- More complete bindings for the gst_video::VideoOverlay interface,
+ especially
+ gst_video::is_video_overlay_prepare_window_handle_message()
+
+Changed
+
+- All references were updated from GitHub to freedesktop.org GitLab
+- Fix various links in the README.md
+- Link to the correct location for the documentation
+- Remove GitLab badge as that only works with gitlab.com currently
+
+Changes in git master for 0.13
+
+Fixed
+
+- gst::tag::Album is the album tag now instead of artist sortname
+
+Added
+
+- Subclassing infrastructure was moved directly into the bindings,
+ making the gst-plugin crate deprecated. This involves many API
+ changes but generally cleans up code and makes it more flexible.
+ Take a look at the gst-plugins-rs crate for various examples.
+
+- Bindings for CapsFeatures and Meta
+- Bindings for
+ ParentBufferMeta,VideoMetaandVideoOverlayCompositionMeta`
+- Bindings for VideoOverlayComposition and VideoOverlayRectangle
+- Bindings for VideoTimeCode
+
+- UniqueFlowCombiner and UniqueAdapter wrappers that make use of the
+ Rust compile-time mutability checks and expose more API in a safe
+ way, and as a side-effect implement Sync and Send now
+
+- More complete bindings for Allocation Query
+- pbutils functions for codec descriptions
+- TagList::iter() for iterating over all tags while getting a single
+ value per tag. The old ::iter_tag_list() function was renamed to
+ ::iter_generic() and still provides access to each value for a tag
+- Bus::iter() and Bus::iter_timed() iterators around the corresponding
+ ::pop\*() functions
+
+- serde serialization of Value can also handle Buffer now
+
+- Extensive comments to all examples with explanations
+- Transmuxing example showing how to use typefind, multiqueue and
+ dynamic pads
+- basic-tutorial-12 was ported and added
+
+Changed
+
+- Rust 1.31 is the minimum supported Rust version now
+- Update to latest gir code generator and glib bindings
+
+- Functions returning e.g. gst::FlowReturn or other “combined” enums
+ were changed to return split enums like
+ Result<gst::FlowSuccess, gst::FlowError> to allow usage of the
+ standard Rust error handling.
+
+- MiniObject subclasses are now newtype wrappers around the underlying
+ GstRc<FooRef> wrapper. This does not change the API in any breaking
+ way for the current usages, but allows MiniObjects to also be
+ implemented in other crates and makes sure rustdoc places the
+ documentation in the right places.
+
+- BinExt extension trait was renamed to GstBinExt to prevent conflicts
+ with gtk::Bin if both are imported
+
+- Buffer::from_slice() can’t possible return None
+
+- Various clippy warnings
+
+
+GStreamer Rust Plugins
+Like the GStreamer Rust bindings, the Rust plugins are now officially
+part of the GStreamer project and are also maintained in the GStreamer
+GitLab.
-Bugs fixed in 1.14
+In the 0.3.x versions this contained infrastructure for writing
+GStreamer plugins in Rust, and a set of plugins.
-More than 800 bugs have been fixed during the development of 1.14.
+In git master that infrastructure was moved to the GLib and GStreamer
+bindings directly, together with many other improvements that were made
+possible by this, so the gst-plugins-rs repository only contains
+GStreamer elements now.
-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.
+Elements included are:
-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.
+- Tutorials plugin: identity, rgb2gray and sinesrc with extensive
+ comments
+- rsaudioecho, a port of the audiofx element
-Stable 1.14 branch
+- rsfilesrc, rsfilesink
-After the 1.14.0 release there will be several 1.14.x bug-fix releases
+- rsflvdemux, a FLV demuxer. Not feature-equivalent with flvdemux yet
+
+- threadshare plugin: ts-appsrc, ts-proxysrc/sink, ts-queue, ts-udpsrc
+ and ts-tcpclientsrc elements that use a fixed number of threads and
+ share them between instances. For more background about these
+ elements see Sebastian’s talk “When adding more threads adds more
+ problems - Thread-sharing between elements in GStreamer” at the
+ GStreamer Conference 2017.
+
+- rshttpsrc, a HTTP source around the hyper/reqwest Rust libraries.
+ Not feature-equivalent with souphttpsrc yet.
+
+- togglerecord, an element that allows to start/stop recording at any
+ time and keeps all audio/video streams in sync.
+
+- mccparse and mccenc, parsers and encoders for the MCC closed caption
+ file format.
+
+Changes to 0.3.X since 0.3.0
+
+- All references were updated from GitHub to freedesktop.org GitLab
+- Fix various links in the README.md
+- Link to the correct location for the documentation
+
+Changes in git master for 0.4
+
+- togglerecord: Switch to parking_lot crate for mutexes/condition
+ variables for lower overhead
+- Merge threadshare plugin here
+- New closedcaption plugin with mccparse and mccenc elements
+- New identity element for the tutorials plugin
+
+- Register plugins statically in tests instead of relying on the
+ plugin loader to find the shared library in a specific place
+
+- Update to the latest API changes in the GLib and GStreamer bindings
+- Update to the latest versions of all crates
+
+
+Build and Dependencies
+
+- The MESON BUILD SYSTEM BUILD IS NOW FEATURE-COMPLETE (*) and it is
+ now the recommended build system on all platforms and also used by
+ Cerbero to build GStreamer on all platforms. The Autotools build is
+ scheduled to be removed in the next cycle. Developers who currently
+ use gst-uninstalled should move to gst-build. The build option
+ naming has been cleaned up and made consistent and there are now
+ feature options to enable/disable plugins and various other features
+ on a case-by-case basis. (*) with the exception of plugin docs which
+ will be handled differently in future
+
+- Symbol export in libraries is now controlled via explicit exports
+ using symbol visibility or export defines where supported, to ensure
+ consistency across all platforms. This also allows libraries to have
+ exports that vary based on detected platform features and configure
+ options as is the case with the GStreamer OpenGL integration library
+ for example. A few symbols that had been exported by accident in
+ earlier versions may no longer be exported. These symbols will not
+ have had declarations in any public header files then though and
+ would not have been usable.
+
+- The GStreamer FFmpeg wrapper plugin (gst-libav) now depends on
+ FFmpeg 4.x and uses the new FFmpeg 4.x API and stopped relying on
+ ancient API that was removed with the FFmpeg 4.x release. This means
+ that it is no longer possible to build this module against an older
+ system-provided FFmpeg 3.x version. Use the internal FFmpeg 4.x copy
+ instead if you build using autotools, or use gst-libav 1.14.x
+ instead which targets the FFmpeg 3.x API and _should_ work fine in
+ combination with a newer GStreamer. It’s difficult for us to support
+ both old and new FFmpeg APIs at the same time, apologies for any
+ inconvenience caused.
+
+- Hardware-accelerated Nvidia video encoder/decoder plugins nvdec and
+ nvenc can be built against CUDA Toolkit versions 9 and 10.0 now. The
+ dynlink interface has been dropped since it’s deprecated in 10.0.
+
+- The (optional) OpenCV requirement has been bumped to >= 3.0.0 and
+ the plugin can also be built against OpenCV 4.x now.
+
+- New sctp plugin based on usrsctp (for WebRTC data channels)
+
+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.
+
+Cerbero has seen a number of improvements:
+
+- Cerbero has been ported to Python 3 and requires Python 3.5 or newer
+ now
+
+- Source tarballs are now protected by checksums in the recipes to
+ guard against download errors and malicious takeover of projects or
+ websites. In addition, downloads are only allowed via secure
+ transports now and plain HTTP, FTP and git:// transports are not
+ allowed anymore.
+
+- There is now a new fetch-bootstrap command which downloads sources
+ required for bootstrapping, with an optional --build-tools-only
+ argument to match the bootstrap --build-tools-only command.
+
+- The bootstrap, build, package and bundle-source commands gained a
+ new --offline switch that ensures that only sources from the cache
+ are used and never downloaded via the network. This is useful in
+ combination with the fetch and fetch-bootstrap commands that acquire
+ sources ahead of time before any build steps are executed. This
+ allows more control over the sources used and when sources are
+ updated, and is particularly useful for build environments that
+ don’t have network access.
+
+- bootstrap --assume-yes will automatically say ‘yes’ to any
+ interactive prompts during the bootstrap stage, such as those from
+ apt-get or yum.
+
+- bootstrap --system-only will only bootstrap the system without build
+ tools.
+
+- Manifest support: The build manifest can be used in continuous
+ integration (CI) systems to fixate the Git revision of certain
+ projects so that all builds of a pipeline are on the same reference.
+ This is used in GStreamer’s gitlab CI for example. It can also be
+ used in order to re-produce a specific build. To set a manifest, you
+ can set manifest = 'my_manifest.xml' in your configuration file, or
+ use the --manifest command line option. The command line option will
+ take precendence over anything specific in the configuration file.
+
+- The new build-deps command can be used to build only the
+ dependencies of a recipe, without the recipe itself.
+
+- new --list-variants command to list available variants
+
+- variants can now be set on the command line via the -v option as a
+ comma-separated list. This overrides any variants set in any
+ configuration files.
+
+- new qt5, intelmsdk and nvidia variants for enabling Qt5 and hardware
+ codec support. See the Enabling Optional Features with Variants
+ section in the Cerbero documentation for more details how to enable
+ and use these variants.
+
+- A new -t / --timestamp command line switch makes commands print
+ timestamps
+
+
+Platform-specific changes and improvements
+
+Android
+
+- toolchain: update compiler to clang and NDKr18. NDK r18 removed the
+ armv5 target and only has Android platforms that target at least
+ armv7 so the armv5 target is not useful anymore.
+
+- The way that GIO modules are named has changed due to upstream GLib
+ natively adding support for loading static GIO modules. This means
+ that any GStreamer application using gnutls for SSL/TLS on the
+ Android or iOS platforms (or any other setup using static libraries)
+ will fail to link looking for the g_io_module_gnutls_load_static()
+ function. The new function name is now
+ g_io_gnutls_load(gpointer data). data can be NULL for a static
+ library. Look at this commit for the necessary change in the
+ examples.
+
+- various build issues on Android have been fixed.
+
+macOS and iOS
+
+- various build issues on iOS have been fixed.
+
+- the minimum required iOS version is now 9.0. The difference in
+ adoption between 8.0 and 9.0 is 0.1% and the bump to 9.0 fixes some
+ build issues.
+
+- The way that GIO modules are named has changed due to upstream GLib
+ natively adding support for loading static GIO modules. This means
+ that any GStreamer application using gnutls for SSL/TLS on the
+ Android or iOS platforms (or any other setup using static libraries)
+ will fail to link looking for the g_io_module_gnutls_load_static()
+ function. The new function name is now
+ g_io_gnutls_load(gpointer data). data can be NULL for a static
+ library. Look at this commit for the necessary change in the
+ examples.
+
+Windows
+
+- The webrtcdsp element is shipped again as part of the Windows binary
+ packages, the build system issue has been resolved.
+
+- ‘Inconsistent DLL linkage’ warnings when building with MSVC have
+ been fixed
+
+- Hardware-accelerated Nvidia video encoder/decoder plugins nvdec and
+ nvenc build on Windows now, also with MSVC and using Meson.
+
+- The ksvideosrc camera capture plugin supports 16-bit grayscale video
+ now
+
+- The wasapisrc audio capture element implements loopback recording
+ from another output device or sink
+
+- wasapisink recover from low buffer levels in shared mode and some
+ exclusive mode fixes
+
+- dshowsrc now implements the GstDeviceMonitor interface
+
+
+Contributors
+
+Aaron Boxer, Aleix Conchillo Flaqué, Alessandro Decina, Alexandru Băluț,
+Alex Ashley, Alexey Chernov, Alicia Boya García, Amit Pandya, Andoni
+Morales Alastruey, Andreas Frisch, Andre McCurdy, Andy Green, Anthony
+Violo, Antoine Jacoutot, Antonio Ospite, Arun Raghavan, Aurelien Jarno,
+Aurélien Zanelli, ayaka, Bananahemic, Bastian Köcher, Branko Subasic,
+Brendan Shanks, Carlos Rafael Giani, Charlie Turner, Christoph Reiter,
+Corentin Noël, Daeseok Youn, Damian Vicino, Dan Kegel, Daniel Drake,
+Daniel Klamt, Danilo Spinella, Dardo D Kleiner, David Ing, David
+Svensson Fors, Devarsh Thakkar, Dimitrios Katsaros, Edward Hervey,
+Emilio Pozuelo Monfort, Enrique Ocaña González, Erlend Eriksen, Ezequiel
+Garcia, Fabien Dessenne, Fabrizio Gennari, Florent Thiéry, Francisco
+Velazquez, Freyr666, Garima Gaur, Gary Bisson, George Kiagiadakis, Georg
+Lippitsch, Georg Ottinger, Geunsik Lim, Göran Jönsson, Guillaume
+Desmottes, H1Gdev, Haihao Xiang, Haihua Hu, Harshad Khedkar, Havard
+Graff, He Junyan, Hoonhee Lee, Hosang Lee, Hyunjun Ko, Ilya Smelykh,
+Ingo Randolf, Iñigo Huguet, Jakub Adam, James Stevenson, Jan Alexander
+Steffens, Jan Schmidt, Jerome Laheurte, Jimmy Ohn, Joakim Johansson,
+Jochen Henneberg, Johan Bjäreholt, John-Mark Bell, John Bassett, John
+Nikolaides, Jonathan Karlsson, Jonny Lamb, Jordan Petridis, Josep Torra,
+Joshua M. Doe, Jos van Egmond, Juan Navarro, Julian Bouzas, Jun Xie,
+Junyan He, Justin Kim, Kai Kang, Kim Tae Soo, Kirill Marinushkin, Kyrylo
+Polezhaiev, Lars Petter Endresen, Linus Svensson, Louis-Francis
+Ratté-Boulianne, Lucas Stach, Luis de Bethencourt, Luz Paz, Lyon Wang,
+Maciej Wolny, Marc-André Lureau, Marc Leeman, Marco Trevisan (Treviño),
+Marcos Kintschner, Marian Mihailescu, Marinus Schraal, Mark Nauwelaerts,
+Marouen Ghodhbane, Martin Kelly, Matej Knopp, Mathieu Duponchelle,
+Matteo Valdina, Matthew Waters, Matthias Fend, memeka, Michael Drake,
+Michael Gruner, Michael Olbrich, Michael Tretter, Miguel Paris, Mike
+Wey, Mikhail Fludkov, Naveen Cherukuri, Nicola Murino, Nicolas Dufresne,
+Niels De Graef, Nirbheek Chauhan, Norbert Wesp, Ognyan Tonchev, Olivier
+Crête, Omar Akkila, Pat DeSantis, Patricia Muscalu, Patrick Radizi,
+Patrik Nilsson, Paul Kocialkowski, Per Forlin, Peter Körner, Peter
+Seiderer, Petr Kulhavy, Philippe Normand, Philippe Renon, Philipp Zabel,
+Pierre Labastie, Piotr Drąg, Roland Jon, Roman Sivriver, Roman Shpuntov,
+Rosen Penev, Russel Winder, Sam Gigliotti, Santiago Carot-Nemesio,
+Sean-Der, Sebastian Dröge, Seungha Yang, Shi Yan, Sjoerd Simons, Snir
+Sheriber, Song Bing, Soon, Thean Siew, Sreerenj Balachandran, Stefan
+Ringel, Stephane Cerveau, Stian Selnes, Suhas Nayak, Takeshi Sato,
+Thiago Santos, Thibault Saunier, Thomas Bluemel, Tianhao Liu,
+Tim-Philipp Müller, Tobias Ronge, Tomasz Andrzejak, Tomislav Tustonić,
+U. Artie Eoff, Ulf Olsson, Varunkumar Allagadapa, Víctor Guzmán, Víctor
+Manuel Jáquez Leal, Vincenzo Bono, Vineeth T M, Vivia Nikolaidou, Wang
+Fei, wangzq, Whoopie, Wim Taymans, Wind Yuan, Wonchul Lee, Xabier
+Rodriguez Calvar, Xavier Claessens, Haihao Xiang, Yacine Bandou,
+Yeongjin Jeong, Yuji Kuwabara, Zeeshan Ali,
+
+… and many others who have contributed bug reports, translations, sent
+suggestions or helped testing.
+
+
+Stable 1.16 branch
+
+After the 1.16.0 release there will be several 1.16.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.
+a bug-fix release usually. The 1.16.x bug-fix releases will be made from
+the git 1.16 branch, which is a stable branch.
-1.14.0
+1.16.0
-1.14.0 is scheduled to be released in early March 2018.
+1.16.0 was released on 19 April 2019.
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.
+- possibly breaking/incompatible changes to properties of wrapped
+ FFmpeg decoders and encoders (see above).
+
+- The way that GIO modules are named has changed due to upstream GLib
+ natively adding support for loading static GIO modules. This means
+ that any GStreamer application using gnutls for SSL/TLS on the
+ Android or iOS platforms (or any other setup using static libraries)
+ will fail to link looking for the g_io_module_gnutls_load_static()
+ function. The new function name is now
+ g_io_gnutls_load(gpointer data). See Android/iOS sections above for
+ further details.
-Schedule for 1.16
+Schedule for 1.18
-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 1.18, and 1.17 will be the
+unstable development version leading up to the stable 1.18 release. The
+development of 1.17/1.18 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 2017 followed by
-several 1.15 pre-releases and the new 1.16 stable release in September.
+The plan for the 1.18 development cycle is yet to be confirmed, but it
+is possible that the next cycle will be a short one in which case
+feature freeze would be perhaps around August 2019 with a new 1.18
+stable release in September.
-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.
+1.18 will be backwards-compatible to the stable 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._
+_contributions from Sebastian Dröge, Guillaume Desmottes, Matthew
+Waters, _ _Thibault Saunier, and Víctor Manuel Jáquez Leal._
_License: CC BY-SA 4.0_
projects / platform / upstream / gstreamer.git / blobdiff