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| -rw-r--r-- | RELEASE | 15 | ||||
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@@ -1,21 +1,25 @@ -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 @@ -30,1165 +34,154 @@ other improvements. 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, @@ -1196,8 +189,6 @@ several 1.15 pre-releases and the new 1.16 stable release in September. ------------------------------------------------------------------------ -_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_ @@ -1,18 +1,15 @@ -This is GStreamer gst-rtsp-server 1.14.0. +This is GStreamer gst-rtsp-server 1.15.0.1. -The GStreamer team is thrilled to announce a new major feature release in the -stable 1.x API series of your favourite cross-platform multimedia framework! +GStreamer 1.15 is the development version leading up to the next major +stable version which will be 1.16. -As always, this release is again packed with new features, bug fixes and -other improvements. - -The 1.14 release series adds new features on top of the 1.12 series and is +The 1.15 development series adds new features on top of the 1.14 series and is part of the API and ABI-stable 1.x release series of the GStreamer multimedia framework. -Full release notes can be found at: +Full release notes will one day be found at: - https://gstreamer.freedesktop.org/releases/1.14/ + https://gstreamer.freedesktop.org/releases/1.16/ Binaries for Android, iOS, Mac OS X and Windows will be provided shortly after the release. diff --git a/configure.ac b/configure.ac index 2d533ea..ac520a9 100644 --- a/configure.ac +++ b/configure.ac @@ -2,7 +2,7 @@ AC_PREREQ(2.69) dnl initialize autoconf dnl when going to/from release please set the nano (fourth number) right ! dnl releases only do Wall, cvs and prerelease does Werror too -AC_INIT([GStreamer RTSP Server Library], [1.14.0], +AC_INIT([GStreamer RTSP Server Library], [1.15.0.1], [http://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer], [gst-rtsp-server]) AG_GST_INIT @@ -53,13 +53,13 @@ dnl 1.2.5 => 205 dnl 1.10.9 (who knows) => 1009 dnl dnl sets GST_LT_LDFLAGS -AS_LIBTOOL(GST, 1400, 0, 1400) +AS_LIBTOOL(GST, 1500, 0, 1500) dnl *** required versions of GStreamer stuff *** -GST_REQ=1.14.0 -GSTPB_REQ=1.14.0 -GSTPG_REQ=1.14.0 -GSTPD_REQ=1.14.0 +GST_REQ=1.15.0.1 +GSTPB_REQ=1.15.0.1 +GSTPG_REQ=1.15.0.1 +GSTPD_REQ=1.15.0.1 dnl *** autotools stuff **** diff --git a/meson.build b/meson.build index 9682e26..775940f 100644 --- a/meson.build +++ b/meson.build @@ -1,5 +1,5 @@ project('gst-rtsp-server', 'c', - version : '1.14.0', + version : '1.15.0.1', meson_version : '>= 0.33.0', default_options : ['warning_level=1', 'buildtype=debugoptimized']) |