-GSTREAMER 1.14 RELEASE NOTES
+GSTREAMER 1.16 RELEASE NOTES
-The GStreamer team is proud to announce a new major feature release in
-the stable 1.x API series of your favourite cross-platform multimedia
-framework!
+GStreamer 1.16 has not been released yet. It is scheduled for release
+around September 2018.
-As always, this release is again packed with new features, bug fixes and
-other improvements.
+1.15.0.1 is the unstable development version that is being developed in
+the git master branch and which will eventually result in 1.16.
+
+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.
-GStreamer 1.14.0 was released on 19 March 2018.
+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.
-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 19 March 2018, 12:00 UTC (log)_
+_Last updated: Tuesday 20 March 2018, 01:30 UTC (log)_
Introduction
Highlights
-- WebRTC support: real-time audio/video streaming to and from web
- browsers
-
-- Experimental support for the next-gen royalty-free AV1 video codec
-
-- Video4Linux: encoding support, stable element names and faster
- device probing
-
-- Support for the Secure Reliable Transport (SRT) video streaming
- protocol
-
-- RTP Forward Error Correction (FEC) support (ULPFEC)
-
-- RTSP 2.0 support in rtspsrc and gst-rtsp-server
-
-- ONVIF audio backchannel support in gst-rtsp-server and rtspsrc
-
-- playbin3 gapless playback and pre-buffering support
-
-- tee, our stream splitter/duplication element, now does allocation
- query aggregation which is important for efficient data handling and
- zero-copy
-
-- 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.
-
-- rtpjitterbuffer fast-start mode and timestamp offset adjustment
- smoothing
-
-- souphttpsrc connection sharing, which allows for connection reuse,
- cookie sharing, etc.
-
-- nvdec: new plugin for hardware-accelerated video decoding using the
- NVIDIA NVDEC API
-
-- Adaptive DASH trick play support
-
-- ipcpipeline: new plugin that allows splitting a pipeline across
- multiple processes
-
-- Major gobject-introspection annotation improvements for large parts
- of the library API
-
-- GStreamer C# bindings have been revived and seen many updates and
- fixes
-
-- The externally maintained GStreamer Rust bindings had many usability
- improvements and cover most of the API now. Coinciding with the 1.14
- release, a new release with the 1.14 API additions is happening.
+- this section will be completed in due course
Major new features and changes
-WebRTC support
-
-There is now basic support for WebRTC in GStreamer in form of a new
-webrtcbin element and a webrtc support library. This allows you to build
-applications that set up connections with and stream to and from other
-WebRTC peers, whilst leveraging all of the usual GStreamer features such
-as hardware-accelerated encoding and decoding, OpenGL integration,
-zero-copy and embedded platform support. And it's easy to build and
-integrate into your application too!
-
-WebRTC enables real-time communication of audio, video and data with web
-browsers and native apps, and it is supported or about to be support by
-recent versions of all major browsers and operating systems.
-
-GStreamer's new WebRTC implementation uses libnice for Interactive
-Connectivity Establishment (ICE) to figure out the best way to
-communicate with other peers, punch holes into firewalls, and traverse
-NATs.
-
-The implementation is not complete, but all the basics are there, and
-the code sticks fairly close to the PeerConnection API. Where
-functionality is missing it should be fairly obvious where it needs to
-go.
+Noteworthy new API
-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.
+- this section will be filled in in due course
New Elements
-- webrtcbin handles the transport aspects of webrtc connections (see
- WebRTC section above for more details)
-
-- New srtsink and srtsrc elements for the Secure Reliable Transport
- (SRT) video streaming protocol, which aims to be easy to use whilst
- striking a new balance between reliability and latency for low
- latency video streaming use cases. More details about SRT and the
- implementation in GStreamer in Olivier's blog post _SRT in
- GStreamer_.
-
-- av1enc and av1dec elements providing experimental support for the
- next-generation royalty free video AV1 codec, alongside Matroska
- support for it.
-
-- hlssink2 is a rewrite of the existing hlssink element, but unlike
- its predecessor hlssink2 takes elementary streams as input and
- handles the muxing to MPEG-TS internally. It also leverages
- splitmuxsink internally to do the splitting. This allows more
- control over the chunk splitting and sizing process and relies less
- on the co-operation of an upstream muxer. Different to the old
- hlssink it also works with pre-encoded streams and does not require
- close interaction with an upstream encoder element.
-
-- audiolatency is a new element for measuring audio latency end-to-end
- and is useful to measure roundtrip latency including both the
- GStreamer-internal latency as well as latency added by external
- components or circuits.
-
-- 'fakevideosink is basically a null sink for video data and very
- similar to fakesink, only that it will answer allocation queries and
- will advertise support for various video-specific things such
- GstVideoMeta, GstVideoCropMeta and GstVideoOverlayCompositionMeta
- like a normal video sink would. This is useful for throughput
- testing and testing the zero-copy path when creating a new pipeline.
-
-- ipcpipeline: new plugin that allows the splitting of a pipeline into
- multiple processes. Usually a GStreamer pipeline runs in a single
- process and parallelism is achieved by distributing workloads using
- multiple threads. This means that all elements in the pipeline have
- access to all the other elements' memory space however, including
- that of any libraries used. For security reasons one might therefore
- want to put sensitive parts of a pipeline such as DRM and decryption
- handling into a separate process to isolate it from the rest of the
- pipeline. This can now be achieved with the new ipcpipeline plugin.
- Check out George's blog post _ipcpipeline: Splitting a GStreamer
- pipeline into multiple processes_ or his lightning talk from last
- year's GStreamer Conference in Prague for all the gory details.
-
-- proxysink and proxysrc are new elements to pass data from one
- pipeline to another within the same process, very similar to the
- existing inter elements, but not limited to raw audio and video
- data. These new proxy elements are very special in how they work
- under the hood, which makes them extremely powerful, but also
- dangerous if not used with care. The reason for this is that it's
- not just data that's passed from sink to src, but these elements
- basically establish a two-way wormhole that passes through queries
- and events in both directions, which means caps negotiation and
- allocation query driven zero-copy can work through this wormhole.
- There are scheduling considerations as well: proxysink forwards
- everything into the proxysrc pipeline directly from the proxysink
- streaming thread. There is a queue element inside proxysrc to
- decouple the source thread from the sink thread, but that queue is
- not unlimited, so it is entirely possible that the proxysink
- pipeline thread gets stuck in the proxysrc pipeline, e.g. when that
- pipeline is paused or stops consuming data for some other reason.
- This means that one should always shut down down the proxysrc
- pipeline before shutting down the proxysink pipeline, for example.
- Or at least take care when shutting down pipelines. Usually this is
- not a problem though, especially not in live pipelines. For more
- information see Nirbheek's blog post _Decoupling GStreamer
- Pipelines_, and also check out out the new ipcpipeline plugin for
- sending data from one process to another process (see above).
-
-- lcms is a new LCMS-based ICC color profile correction element
-
-- openmptdec is a new OpenMPT-based decoder for module music formats,
- such as S3M, MOD, XM, IT. It is built on top of a new
- GstNonstreamAudioDecoder base class which aims to unify handling of
- files which do not operate a streaming model. The wildmidi plugin
- has also been revived and is also implemented on top of this new
- base class.
-
-- The curl plugin has gained a new curlhttpsrc element, which is
- useful for testing HTTP protocol version 2.0 amongst other things.
-
-- The msdk plugin has gained a MPEG-2 video decoder(msdkmpeg2dec), VP8
- decoder(msdkvp8dec) and a VC1/WMV decoder(msdkvc1dec)
+- this section will be filled in in due course
-Noteworthy new API
+New element features and additions
-- GstPromise provides future/promise-like functionality. This is used
- in the GStreamer WebRTC implementation.
-
-- GstReferenceTimestampMeta is a new meta that allows you to attach
- additional reference timestamps to a buffer. These timestamps don't
- have to relate to the pipeline clock in any way. Examples of this
- could be an NTP timestamp when the media was captured, a frame
- counter on the capture side or the (local) UNIX timestamp when the
- media was captured. The decklink elements make use of this.
-
-- GstVideoRegionOfInterestMeta: it's now possible to attach generic
- free-form element-specific parameters to a region of interest meta,
- for example to tell a downstream encoder to use certain codec
- parameters for a certain region.
-
-- gst_bus_get_pollfd can be used to obtain a file descriptor for the
- bus that can be poll()-ed on for new messages. This is useful for
- integration with non-GLib event loops.
-
-- gst_get_main_executable_path() can be used by wrapper plugins that
- need to find things in the directory where the application
- executable is located. In the same vein,
- GST_PLUGIN_DEPENDENCY_FLAG_PATHS_ARE_RELATIVE_TO_EXE can be used to
- signal that plugin dependency paths are relative to the main
- executable.
-
-- pad templates can be told about the GType of the pad subclass of the
- pad via newly-added GstPadTemplate API API or the
- gst_element_class_add_static_pad_template_with_gtype() convenience
- function. gst-inspect-1.0 will use this information to print pad
- properties.
-
-- new convenience functions to iterate over element pads without using
- the GstIterator API: gst_element_foreach_pad(),
- gst_element_foreach_src_pad(), and gst_element_foreach_sink_pad().
-
-- GstBaseSrc and appsrc have gained support for buffer lists:
- GstBaseSrc subclasses can use gst_base_src_submit_buffer_list(), and
- applications can use gst_app_src_push_buffer_list() to push a buffer
- list into appsrc.
-
-- The GstHarness unit test harness has a couple of new convenience
- functions to retrieve all pending data in the harness in form of a
- single chunk of memory.
-
-- GstAudioStreamAlign is a new helper object for audio elements that
- handles discontinuity detection and sample alignment. It will align
- samples after the previous buffer's samples, but keep track of the
- divergence between buffer timestamps and sample position (jitter).
- If it exceeds a configurable threshold the alignment will be reset.
- This simply factors out code that was duplicated in a number of
- elements into a common helper API.
-
-- The GstVideoEncoder base class implements Quality of Service (QoS)
- now. This is disabled by default and must be opted in by setting the
- "qos" property, which will make the base class gather statistics
- about the real-time performance of the pipeline from downstream
- elements (usually sinks that sync the pipeline clock). Subclasses
- can then make use of this by checking whether input frames are late
- already using gst_video_encoder_get_max_encode_time() If late, they
- can just drop them and skip encoding in the hope that the pipeline
- will catch up.
-
-- The GstVideoOverlay interface gained a few helper functions for
- installing and handling a "render-rectangle" property on elements
- that implement this interface, so that this functionality can also
- be used from the command line for testing and debugging purposes.
- The property wasn't added to the interface itself as that would
- require all implementors to provide it which would not be
- backwards-compatible.
-
-- A new base class, GstNonstreamAudioDecoder for non-stream audio
- decoders was added to gst-plugins-bad. This base-class is meant to
- be used for audio decoders that require the whole stream to be
- loaded first before decoding can start. Examples of this are module
- formats (MOD/S3M/XM/IT/etc), C64 SID tunes, video console music
- files (GYM/VGM/etc), MIDI files and others. The new openmptdec
- element is based on this.
-
-- Full list of API new in 1.14:
-- GStreamer core API new in 1.14
-- GStreamer base library API new in 1.14
-- gst-plugins-base libraries API new in 1.14
-- gst-plugins-bad: no list, mostly GstWebRTC library and new
- non-stream audio decoder base class.
-
-New RTP features and improvements
-
-- rtpulpfecenc and rtpulpfecdec are new elements that implement
- Generic Forward Error Correction (FEC) using Uneven Level Protection
- (ULP) as described in RFC 5109. This can be used to protect against
- certain types of (non-bursty) packet loss, and important packets
- such as those containing codec configuration data or key frames can
- be protected with higher redundancy. Equally, packets that are not
- particularly important can be given low priority or not be protected
- at all. If packets are lost, the receiver can then hopefully restore
- the lost packet(s) from the surrounding packets which were received.
- This is an alternative to, or rather complementary to, dealing with
- packet loss using _retransmission (rtx)_. GStreamer has had
- retransmission support for a long time, but Forward Error Correction
- allows for different trade-offs: The advantage of Forward Error
- Correction is that it doesn't add latency, whereas retransmission
- requires at least one more roundtrip to request and hopefully
- receive lost packets; Forward Error Correction increases the
- required bandwidth however, even in situations where there is no
- packet loss at all, so one will typically want to fine-tune the
- overhead and mechanisms used based on the characteristics of the
- link at the time.
-
-- New _Redundant Audio Data (RED)_ encoders and decoders for RTP as
- per RFC 2198 are also provided (rtpredenc and rtpreddec), mostly for
- chrome webrtc compatibility, as chrome will wrap ULPFEC-protected
- streams in RED packets, and such streams need to be wrapped and
- unwrapped in order to use ULPFEC with chrome.
-
-- a few new buffer flags for FEC support:
- GST_BUFFER_FLAG_NON_DROPPABLE can be used to mark important buffers,
- e.g. to flag RTP packets carrying keyframes or codec setup data for
- RTP Forward Error Correction purposes, or to prevent still video
- frames from being dropped by elements due to QoS. There already is a
- GST_BUFFER_FLAG_DROPPABLE. GST_RTP_BUFFER_FLAG_REDUNDANT is used to
- signal internally that a packet represents a redundant RTP packet
- and used in rtpstorage to hold back the packet and use it only for
- recovery from packet loss. Further work is still needed in
- payloaders to make use of these.
-
-- rtpbin now has an option for increasing timestamp offsets gradually:
- Sudden large changes to the internal ts_offset may cause timestamps
- to move backwards and may also cause visible glitches in media
- playback. The new "max-ts-offset-adjustment" and "max-ts-offset"
- properties let the application control the rate to apply changes to
- ts_offset. There have also been some EOS/BYE handling improvements
- in rtpbin.
-
-- rtpjitterbuffer has a new fast start mode: in many scenarios the
- jitter buffer will have to wait for the full configured latency
- before it can start outputting packets. The reason for that is that
- it often can't know what the sequence number of the first expected
- RTP packet is, so it can't know whether a packet earlier than the
- earliest packet received will still arrive in future. This behaviour
- can now be bypassed by setting the "faststart-min-packets" property
- to the number of consecutive packets needed to start, and the jitter
- buffer will start output packets as soon as it has N consecutive
- packets queued internally. This is particularly useful to get a
- first video frame decoded and rendered as quickly as possible.
-
-- rtpL8pay and rtpL8depay provide RTP payloading and depayloading for
- 8-bit raw audio
-
-New element features
-
-- playbin3 has gained support or gapless playback via the
- "about-to-finish" signal where users can set the uri for the next
- item to play. For non-live streams this will be emitted as soon as
- the first uri has finished downloading, so with sufficiently large
- buffers it is now possible to pre-buffer the next item well ahead of
- time (unlike playbin where there would not be a lot of time between
- "about-to-finish" emission and the end of the stream). If the stream
- format of the next stream is the same as that of the previous
- stream, the data will be concatenated via the concat element.
- Whether this will result in true gaplessness depends on the
- container format and codecs used, there might still be codec-related
- gaps between streams with some codecs.
-
-- tee now does allocation query aggregation, which is important for
- zero-copy and efficient data handling, especially for video. Those
- who want to drop allocation queries on purpose can use the identity
- element's new "drop-allocation" property for that instead.
-
-- audioconvert now has a "mix-matrix" property, which obsoletes the
- audiomixmatrix element. There's also mix matrix support in the audio
- conversion and channel mixing API.
-
-- x264enc: new "insert-vui" property to disable VUI (Video Usability
- Information) parameter insertion into the stream, which allows
- creation of streams that are compatible with certain legacy hardware
- decoders that will refuse to decode in certain combinations of
- resolution and VUI parameters; the max. allowed number of B-frames
- was also increased from 4 to 16.
-
-- dvdlpcmdec: has gained support for Blu-Ray audio LPCM.
-
-- appsrc has gained support for buffer lists (see above) and also seen
- some other performance improvements.
-
-- flvmux has been ported to the GstAggregator base class which means
- it can work in defined-latency mode with live input sources and
- continue streaming if one of the inputs stops producing data.
-
-- jpegenc has gained a "snapshot" property just like pngenc to make it
- easier to output just a single encoded frame.
-
-- jpegdec will now handle interlaced MJPEG streams properly and also
- handles frames without an End of Image marker better.
-
-- v4l2: There are now video encoders for VP8, VP9, MPEG4, and H263.
- The v4l2 video decoder handles dynamic resolution changes, and the
- video4linux device provider now does much faster device probing. The
- plugin also no longer uses the libv4l2 library by default, as it has
- prevented a lot of interesting use cases like CREATE_BUFS, DMABuf,
- usage of TRY_FMT. As the libv4l2 library is totally inactive and not
- really maintained, we decided to disable it. This might affect a
- small number of cheap/old webcams with custom vendor formats for
- which we do not provide conversion in GStreamer. It is possible to
- re-enable support for libv4l2 at run-time however, by setting the
- environment variable GST_V4L2_USE_LIBV4L2=1.
-
-- rtspsrc now has support for RTSP protocol version 2.0 as well as
- ONVIF audio backchannels (see below for more details). It also
- sports a new "accept-certificate" signal for "manually" checking a
- TLS certificate for validity. It now also prints RTSP/SDP messages
- to the gstreamer debug log instead of stdout.
-
-- shout2send now uses non-blocking I/O and has a configurable network
- operations timeout.
-
-- splitmuxsink has gained a "split-now" action signal and new
- "alignment-threshold" and "use-robust-muxing" properties. If robust
- muxing is enabled, it will check and set the muxer's reserved space
- properties if present. This is primarily for use with mp4mux's
- robust muxing mode.
-
-- qtmux has a new _prefill recording mode_ which sets up a moov header
- with the correct sample positions beforehand, which then allows
- software like Adobe Premiere and FinalCut Pro to import the files
- while they are still being written to. This only works with constant
- framerate I-frame only streams, and for now only support for ProRes
- video and raw audio is implemented. Adding support for additional
- codecs is just a matter of defining appropriate maximum frame sizes
- though.
-
-- qtmux also supports writing of svmi atoms with stereoscopic video
- information now. Trak timescales can be configured on a per-stream
- basis using the "trak-timescale" property on the sink pads. Various
- new formats can be muxed: MPEG layer 1 and 2, AC3 and Opus, as well
- as PNG and VP9.
-
-- souphttpsrc now does connection sharing by default: it shares its
- SoupSession with other elements in the same pipeline via a
- GstContext if possible (session-wide settings are all the defaults).
- This allows for connection reuse, cookie sharing, etc. Applications
- can also force a context to use. In other news, HTTP headers
- received from the server are posted as element messages on the bus
- now for easier diagnostics, and it's also possible now to use other
- types of proxy servers such as SOCKS4 or SOCKS5 proxies, support for
- which is implemented directly in gio. Before only HTTP proxies were
- allowed.
-
-- qtmux, mp4mux and matroskamux will now refuse caps changes of input
- streams at runtime. This isn't really supported with these
- containers (or would have to be implemented differently with a
- considerable effort) and doesn't produce valid and spec-compliant
- files that will play everywhere. So if you can't guarantee that the
- input caps won't change, use a container format that does support on
- the fly caps changes for a stream such as MPEG-TS or use
- splitmuxsink which can start a new file when the caps change. What
- would happen before is that e.g. rtph264depay or rtph265depay would
- simply send new SPS/PPS inband even for AVC format, which would then
- get muxed into the container as if nothing changed. Some decoders
- will handle this just fine, but that's often more luck than by
- design. In any case, it's not right, so we disallow it now.
-
-- matroskamux has Table of Content (TOC) support now (chapters etc.)
- and matroskademux TOC support has been improved. matroskademux has
- also seen seeking improvements searching for the right cluster and
- position.
-
-- videocrop now uses GstVideoCropMeta if downstream supports it, which
- means cropping can be handled more efficiently without any copying.
-
-- compositor now has support for _crossfade blending_, which can be
- used via the new "crossfade-ratio" property on the sink pads.
-
-- The avwait element has a new "end-timecode" property and posts
- "avwait-status" element messages now whenever avwait starts or stops
- passing through data (e.g. because target-timecode and end-timecode
- respectively have been reached).
-
-- h265parse and h265parse will try harder to make upstream output the
- same caps as downstream requires or prefers, thus avoiding
- unnecessary conversion. The parsers also expose chroma format and
- bit depth in the caps now.
-
-- The dtls elements now longer rely on or require the application to
- run a GLib main loop that iterates the default main context
- (GStreamer plugins should never rely on the application running a
- GLib main loop).
-
-- openh264enc allows to change the encoding bitrate dynamically at
- runtime now
-
-- nvdec is a new plugin for hardware-accelerated video decoding using
- the NVIDIA NVDEC API (which replaces the old VDPAU API which is no
- longer supported by NVIDIA)
-
-- The NVIDIA NVENC hardware-accelerated video encoders now support
- dynamic bitrate and preset reconfiguration and support the I420
- 4:2:0 video format. It's also possible to configure the gop size via
- the new "gop-size" property.
-
-- The MPEG-TS muxer and demuxer (tsmux, tsdemux) now have support for
- JPEG2000
-
-- openjpegdec and jpeg2000parse support 2-component images now (gray
- with alpha), and jpeg2000parse has gained limited support for
- conversion between JPEG2000 stream-formats. (JP2, J2C, JPC) and also
- extracts more details such as colorimetry, interlace-mode,
- field-order, multiview-mode and chroma siting.
-
-- The decklink plugin for Blackmagic capture and playback cards have
- seen numerous improvements:
-
-- decklinkaudiosrc and decklinkvideosrc now put hardware reference
- timestamp on buffers in form of GstReferenceTimestampMetas.
- This can be useful to know on multi-channel cards which frames from
- different channels were captured at the same time.
-
-- decklinkvideosink has gained support for Decklink hardware keying
- with two new properties ("keyer-mode" and "keyer-level") to control
- the built-in hardware keyer of Decklink cards.
-
-- decklinkaudiosink has been re-implemented around GstBaseSink instead
- of the GstAudioBaseSink base class, since the Decklink APIs don't
- fit very well with the GstAudioBaseSink APIs, which used to cause
- various problems due to inaccuracies in the clock calculations.
- Problems were audio drop-outs and A/V sync going wrong after
- pausing/seeking.
-
-- support for more than 16 devices, without any artificial limit
-
-- work continued on the msdk plugin for Intel's Media SDK which
- enables hardware-accelerated video encoding and decoding on Intel
- graphics hardware on Windows or Linux. Added the video memory,
- buffer pool, and context/session sharing support which helps to
- improve the performance and resource utilization. Rendernode support
- is in place which helps to avoid the constraint of having a running
- graphics server as DRM-Master. Encoders are exposing a number rate
- control algorithms now. More encoder tuning options like
- trellis-quantiztion (h264), slice size control (h264), B-pyramid
- prediction(h264), MB-level bitrate control, frame partitioning and
- adaptive I/B frame insertion were added, and more pixel formats and
- video codecs are supported now. The encoder now also handles
- force-key-unit events and can insert frame-packing SEIs for
- side-by-side and top-bottom stereoscopic 3D video.
-
-- dashdemux can now do adaptive trick play of certain types of DASH
- streams, meaning it can do fast-forward/fast-rewind of normal (non-I
- frame only) streams even at high speeds without saturating network
- bandwidth or exceeding decoder capabilities. It will keep statistics
- and skip keyframes or fragments as needed. See Sebastian's blog post
- _DASH trick-mode playback in GStreamer_ for more details. It also
- supports webvtt subtitle streams now and has seen improvements when
- seeking in live streams.
-
-- kmssink has seen lots of fixes and improvements in this cycle,
- including:
-
-- Raspberry Pi (vc4) and Xilinx DRM driver support
-
-- new "render-rectangle" property that can be used from the command
- line as well as "display-width" and "display-height", and
- "can-scale" properties
-
-- GstVideoCropMeta support
+- this section will be filled in in due course
Plugin and library moves
-MPEG-1 audio (mp1, mp2, mp3) decoders and encoders moved to -good
-
-Following the expiration of the last remaining mp3 patents in most
-jurisdictions, and the termination of the mp3 licensing program, as well
-as the decision by certain distros to officially start shipping full mp3
-decoding and encoding support, these plugins should now no longer be
-problematic for most distributors and have therefore been moved from
--ugly and -bad to gst-plugins-good. Distributors can still disable these
-plugins if desired.
-
-In particular these are:
-
-- mpg123audiodec: an mp1/mp2/mp3 audio decoder using libmpg123
-- lamemp3enc: an mp3 encoder using LAME
-- twolamemp2enc: an mp2 encoder using TwoLAME
-
-GstAggregator moved from -bad to core
-
-GstAggregator has been moved from gst-plugins-bad to the base library in
-GStreamer and is now stable API.
-
-GstAggregator is a new base class for mixers and muxers that have to
-handle multiple input pads and aggregate streams into one output stream.
-It improves upon the existing GstCollectPads API in that it is a proper
-base class which was also designed with live streaming in mind.
-GstAggregator subclasses will operate in a mode with defined latency if
-any of the inputs are live streams. This ensures that the pipeline won't
-stall if any of the inputs stop producing data, and that the configured
-maximum latency is never exceeded.
-
-GstAudioAggregator, audiomixer and audiointerleave moved from -bad to -base
-
-GstAudioAggregator is a new base class for raw audio mixers and muxers
-and is based on GstAggregator (see above). It provides defined-latency
-mixing of raw audio inputs and ensures that the pipeline won't stall
-even if one of the input streams stops producing data.
-
-As part of the move to stabilise the API there were some last-minute API
-changes and clean-ups, but those should mostly affect internal elements.
-
-It is used by the audiomixer element, which is a replacement for
-'adder', which did not handle live inputs very well and did not align
-input streams according to running time. audiomixer should behave much
-better in that respect and generally behave as one would expected in
-most scenarios.
-
-Similarly, audiointerleave replaces the 'interleave' element which did
-not handle live inputs or non-aligned inputs very robustly.
-
-GstAudioAggregator and its subclases have gained support for input
-format conversion, which does not include sample rate conversion though
-as that would add additional latency. Furthermore, GAP events are now
-handled correctly.
-
-We hope to move the video equivalents (GstVideoAggregator and
-compositor) to -base in the next cycle, i.e. for 1.16.
-
-GStreamer OpenGL integration library and plugin moved from -bad to -base
-
-The GStreamer OpenGL integration library and opengl plugin have moved
-from gst-plugins-bad to -base and are now part of the stable API canon.
-Not all OpenGL elements have been moved; a few had to be left behind in
-gst-plugins-bad in the new openglmixers plugin, because they depend on
-the GstVideoAggregator base class which we were not able to move in this
-cycle. We hope to reunite these elements with the rest of their family
-for 1.16 though.
-
-This is quite a milestone, thanks to everyone who worked to make this
-happen!
-
-Qt QML and GTK plugins moved from -bad to -good
-
-The Qt QML-based qmlgl plugin has moved to -good and provides a
-qmlglsink video sink element as well as a qmlglsrc element. qmlglsink
-renders video into a QQuickItem, and qmlglsrc captures a window from a
-QML view and feeds it as video into a pipeline for further processing.
-Both elements leverage GStreamer's OpenGL integration. In addition to
-the move to -good the following features were added:
-
-- A proxy object is now used for thread-safe access to the QML widget
- which prevents crashes in corner case scenarios: QML can destroy the
- video widget at any time, so without this we might be left with a
- dangling pointer.
-
-- EGL is now supported with the X11 backend, which works e.g. on
- Freescale imx6
-
-The GTK+ plugin has also moved from -bad to -good. It includes gtksink
-and gtkglsink which both render video into a GtkWidget. gtksink uses
-Cairo for rendering the video, which will work everywhere in all
-scenarios but involves an extra memory copy, whereas gtkglsink fully
-leverages GStreamer's OpenGL integration, but might not work properly in
-all scenarios, e.g. where the OpenGL driver does not properly support
-multiple sharing contexts in different threads; on Linux Nouveau is
-known to be broken in this respect, whilst NVIDIA's proprietary drivers
-and most other drivers generally work fine, and the experience with
-Intel's driver seems to be mixed; some proprietary embedded Linux
-drivers don't work; macOS works).
-
-GstPhysMemoryAllocator interface moved from -bad to -base
-
-GstPhysMemoryAllocator is a marker interface for allocators with
-physical address backed memory.
+- this section will be filled in in due course
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.
+- this section will be filled in in due course
-- 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.
-API removals
+Miscellaneous API additions
-- some MPEG video parser API in the API unstable codecutils library in
- gst-plugins-bad was removed after having been deprecated for 5
- years.
+- this section will be filled in in due course
+GstPlayer
-Miscellaneous changes
+- this section will be filled in in due course
-- 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.
-OpenGL integration
-
-- The GStreamer OpenGL integration library has moved to
- gst-plugins-base and is now part of our stable API.
+Miscellaneous changes
-- new MESA3D GBM BACKEND. On devices with working libdrm support, it
- is possible to use Mesa3D's GBM library to set up an EGL context
- directly on top of KMS. This makes it possible to use the GStreamer
- OpenGL elements without a windowing system if a libdrm- and
- Mesa3D-supported GPU is present.
+- this section will be filled in in due course
-- Prefer wayland display over X11: As most Wayland compositors support
- XWayland, the X11 backend would get selected.
+OpenGL integration
-- gldownload can export dmabufs now, and glupload will advertise
- dmabuf as caps feature.
+- this section will be filled in in due course
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 user can control how
- the delay applied to delayed packets is distributed: This is either
- the uniform distribution (as before) or the normal distribution; in
- addition there is also the gamma distribution which simulates the
- delay on wifi networks better.
+- this section will be filled in in due course
Tools
-- gst-inspect-1.0 now prints pad properties for elements that have pad
- subclasses with special properties, such as compositor or
- audiomixer. This only works for elements that use the newly-added
- GstPadTemplate API API or the
- gst_element_class_add_static_pad_template_with_gtype() convenience
- function to tell GStreamer about the special pad subclass.
-
-- gst-launch-1.0 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
+- this section will be filled in in due course
GStreamer RTSP server
-- Initial support for RTSP protocol version 2.0 was added, which is to
- the best of our knowledge the first RTSP 2.0 implementation ever!
-
-- 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.
+- this section will be filled in in due course
GStreamer VAAPI
-- Improve DMABuf's usage, both upstream and dowstream, and
- memory:DMABuf caps feature is also negotiated when the dmabuf-based
- buffer cannot be mapped onto user-space.
-
-- VA initialization was fixed when it is used in headless systems.
-
-- VA display sharing, through GstContext, among the pipeline, has been
- improved, adding the possibility to the application share its VA
- display (external display) via gst.vaapi.app.Display context.
-
-- VA display cache was removed.
-
-- libva's log messages are now redirected into the GStreamer log
- handler.
-
-- Decoders improved their upstream re-negotiation by avoiding to
- re-instantiate the internal decoder if stream caps are compatible
- with the previous one.
-
-- When downstream doesn't support GstVideoMeta and the decoded frames
- don't have standard strides, they are copied onto system
- memory-based buffers.
-
-- H.264 decoder has a low-latency property, for live streams which
- doesn't conform the H.264 specification but still it is required to
- push the frames to downstream as soon as possible.
-
-- As part of the Google Summer of Code 2017 the H.264 decoder drops
- MVC and SVC frames when base-only property is enabled.
-
-- Added support for libva-2.0 (VA-API 1.0).
-
-- H.264 and H.265 encoders handle Region-Of-Interest metas by adding a
- delta-qp for every rectangle within the frame specified by those
- metas.
-
-- Encoders for H.264 and H.265 set the media profile by the downstream
- caps.
-
-- H.264 encoder inserts an AU delimiter for each encoded frame when
- aud property is enabled (it is only available for certain drivers
- and platforms).
-
-- H.264 encoder supports for P and B hierarchical prediction modes.
-
-- All encoders handles a quality-level property, which is a number
- from 1 to 8, where a lower number means higher quality, but slower
- processing, and vice-versa.
-
-- VP8 and VP9 encoders support constant bit-rate mode (CBR).
-
-- VP8, VP9 and H.265 encoders support variable bit-rate mode (VBR).
-
-- Resurrected GstGLUploadTextureMeta handling for EGL backends.
-
-- H.265 encoder can configure its number of reference frames via the
- refs property.
-
-- Add H.264 encoder mbbrc property, which controls the macro-block
- bitrate as auto, on or off.
-
-- Add H.264 encoder temporal-levels property, to select the number of
- temporal levels to be included.
-
-- Add to H.264 and H.265 encoders the properties qp-ip and qp-ib, to
- handle the QP (quality parameter) difference between the I and P
- frames, and the I and B frames, respectively.
-
-- vaapisink was demoted to marginal rank on Wayland because COGL
- cannot display YUV surfaces.
+- this section will be filled in in due course
GStreamer Editing Services and NLE
-- Handle crossfade in complex scenarios by using the new
- compositorpad::crossfade-ratio property
-
-- Add API allowing to stop using proxies for clips in the timeline
-
-- Allow management of none square pixel aspect ratios by allowing
- application to deal with them in the way they want
-
-- Misc fixes around the timeline editing API
+- this section will be filled in in due course
GStreamer validate
-- Handle running scenarios on live pipelines (in the "content sense",
- not the GStreamer one)
-
-- Implement RTSP support with a basic server based on gst-rtsp-server,
- and add RTSP 1.0 and 2.0 integration tests
-
-- Implement a plugin that allows users to implement configurable
- tests. It currently can check if a particular element is added a
- configurable number of time in the pipeline. In the future that
- plugin should allow us to implement specific tests of any kind in a
- descriptive way
-
-- Add a verbosity configuration which behaves in a similare way as the
- gst-launch-1.0 verbose flags allowing the informations to be
- outputed on any running pipeline when enabling GstValidate.
-
-- Misc optimization in the launcher, making the tests run much faster.
-
-
-GStreamer C# bindings
-
-- Port to the meson build system, autotools support has been removed
-
-- Use a new GlibSharp version, set as a meson subproject
-
-- Update wrapped API to GStreamer 1.14
+- this section will be filled in in due course
-- Removed the need for "glue" code
-- Provide a nuget
+GStreamer Python Bindings
-- Misc API fixes
+- this section will be filled in in due course
Build and Dependencies
-- 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.
-
-- 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.
+- this section will be filled in in due course
Platform-specific improvements
Android
-- ahcsrc (Android camera source) does autofocus now
+- this section will be filled in in due course
macOS and iOS
-- this section will be filled in shortly {FIXME!}
+- this section will be filled in in due course
Windows
-- 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:
-
-- support for more than 2 channels
-
-- a new "low-latency" property to enable low-latency operation (which
- should always be safe to enable)
-
-- 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.
-
-- set realtime thread priority to avoid glitches
-
-- 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.
-
-- 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
-
-- 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.
+- this section will be filled in in due course
Contributors
-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,
+- this section will be filled in in due course
... and many others who have contributed bug reports, translations, sent
suggestions or helped testing.
-Bugs fixed in 1.14
+Bugs fixed in 1.16
+
+- this section will be filled in in due course
-More than 800 bugs have been fixed during the development of 1.14.
+More than XXX bugs have been fixed during the development of 1.16.
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.
+stable 1.16 branch and fixed there as well, all fixes that ended up in
+the 1.16 branch are also included in 1.16.
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.
-Stable 1.14 branch
+Stable 1.16 branch
-After the 1.14.0 release there will be several 1.14.x bug-fix releases
+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.
-
-1.14.0
-
-1.14.0 was released on 19 March 2018.
-
-1.14.1
+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.
-The first 1.14 bug-fix release (1.14.1) is scheduled to be released
-around the end of March or beginning of April.
+1.16.0
-This release only contains bugfixes and it should be safe to update from
-1.14.0.
+1.16.0 is scheduled to be released around September 2018.
Known Issues
-- The webrtcdsp element (which is unrelated to the newly-landed
- GStreamer webrtc support) is currently not shipped as part of the
+- The webrtcdsp element is currently not shipped as part of the
Windows binary packages due to a build system issue.
-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.
The plan for the 1.16 development cycle is yet to be confirmed, but it
-is expected that feature freeze will be around August 2018 followed by
+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.
1.16 will be backwards-compatible to the stable 1.14, 1.12, 1.10, 1.8,
------------------------------------------------------------------------
-_These release notes have been prepared by Tim-Philipp Müller with_
-_contributions from Sebastian Dröge, Sreerenj Balachandran, Thibault
-Saunier_ _and Víctor Manuel Jáquez Leal._
+_These release notes have been prepared by Tim-Philipp Müller._
_License: CC BY-SA 4.0_
projects / platform / upstream / gstreamer.git / commitdiff