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1203 lines
54 KiB
Text
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GSTREAMER 1.14 RELEASE NOTES
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The GStreamer team is proud to announce a new major feature release in
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the stable 1.x API series of your favourite cross-platform multimedia
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framework!
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As always, this release is again packed with new features, bug fixes and
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other improvements.
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GStreamer 1.14.0 was released on 19 March 2018.
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See https://gstreamer.freedesktop.org/releases/1.14/ for the latest
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version of this document.
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_Last updated: Monday 19 March 2018, 12:00 UTC (log)_
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Introduction
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The GStreamer team is proud to announce a new major feature release in
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the stable 1.x API series of your favourite cross-platform multimedia
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framework!
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As always, this release is again packed with new features, bug fixes and
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other improvements.
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Highlights
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- WebRTC support: real-time audio/video streaming to and from web
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browsers
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- Experimental support for the next-gen royalty-free AV1 video codec
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- Video4Linux: encoding support, stable element names and faster
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device probing
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- Support for the Secure Reliable Transport (SRT) video streaming
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protocol
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- RTP Forward Error Correction (FEC) support (ULPFEC)
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- RTSP 2.0 support in rtspsrc and gst-rtsp-server
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- ONVIF audio backchannel support in gst-rtsp-server and rtspsrc
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- playbin3 gapless playback and pre-buffering support
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- tee, our stream splitter/duplication element, now does allocation
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query aggregation which is important for efficient data handling and
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zero-copy
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- QuickTime muxer has a new prefill recording mode that allows file
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import in Adobe Premiere and FinalCut Pro while the file is still
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being written.
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- rtpjitterbuffer fast-start mode and timestamp offset adjustment
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smoothing
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- souphttpsrc connection sharing, which allows for connection reuse,
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cookie sharing, etc.
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- nvdec: new plugin for hardware-accelerated video decoding using the
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NVIDIA NVDEC API
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- Adaptive DASH trick play support
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- ipcpipeline: new plugin that allows splitting a pipeline across
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multiple processes
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- Major gobject-introspection annotation improvements for large parts
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of the library API
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- GStreamer C# bindings have been revived and seen many updates and
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fixes
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- The externally maintained GStreamer Rust bindings had many usability
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improvements and cover most of the API now. Coinciding with the 1.14
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release, a new release with the 1.14 API additions is happening.
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Major new features and changes
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WebRTC support
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There is now basic support for WebRTC in GStreamer in form of a new
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webrtcbin element and a webrtc support library. This allows you to build
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applications that set up connections with and stream to and from other
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WebRTC peers, whilst leveraging all of the usual GStreamer features such
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as hardware-accelerated encoding and decoding, OpenGL integration,
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zero-copy and embedded platform support. And it's easy to build and
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integrate into your application too!
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WebRTC enables real-time communication of audio, video and data with web
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browsers and native apps, and it is supported or about to be support by
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recent versions of all major browsers and operating systems.
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GStreamer's new WebRTC implementation uses libnice for Interactive
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Connectivity Establishment (ICE) to figure out the best way to
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communicate with other peers, punch holes into firewalls, and traverse
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NATs.
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The implementation is not complete, but all the basics are there, and
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the code sticks fairly close to the PeerConnection API. Where
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functionality is missing it should be fairly obvious where it needs to
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go.
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For more details, background and example code, check out Nirbheek's blog
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post _GStreamer has grown a WebRTC implementation_, as well as Matthew's
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_GStreamer WebRTC_ talk from last year's GStreamer Conference in Prague.
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New Elements
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- webrtcbin handles the transport aspects of webrtc connections (see
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WebRTC section above for more details)
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- New srtsink and srtsrc elements for the Secure Reliable Transport
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(SRT) video streaming protocol, which aims to be easy to use whilst
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striking a new balance between reliability and latency for low
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latency video streaming use cases. More details about SRT and the
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implementation in GStreamer in Olivier's blog post _SRT in
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GStreamer_.
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- av1enc and av1dec elements providing experimental support for the
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next-generation royalty free video AV1 codec, alongside Matroska
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support for it.
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- hlssink2 is a rewrite of the existing hlssink element, but unlike
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its predecessor hlssink2 takes elementary streams as input and
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handles the muxing to MPEG-TS internally. It also leverages
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splitmuxsink internally to do the splitting. This allows more
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control over the chunk splitting and sizing process and relies less
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on the co-operation of an upstream muxer. Different to the old
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hlssink it also works with pre-encoded streams and does not require
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close interaction with an upstream encoder element.
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- audiolatency is a new element for measuring audio latency end-to-end
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and is useful to measure roundtrip latency including both the
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GStreamer-internal latency as well as latency added by external
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components or circuits.
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- 'fakevideosink is basically a null sink for video data and very
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similar to fakesink, only that it will answer allocation queries and
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will advertise support for various video-specific things such
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GstVideoMeta, GstVideoCropMeta and GstVideoOverlayCompositionMeta
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like a normal video sink would. This is useful for throughput
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testing and testing the zero-copy path when creating a new pipeline.
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- ipcpipeline: new plugin that allows the splitting of a pipeline into
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multiple processes. Usually a GStreamer pipeline runs in a single
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process and parallelism is achieved by distributing workloads using
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multiple threads. This means that all elements in the pipeline have
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access to all the other elements' memory space however, including
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that of any libraries used. For security reasons one might therefore
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want to put sensitive parts of a pipeline such as DRM and decryption
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handling into a separate process to isolate it from the rest of the
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pipeline. This can now be achieved with the new ipcpipeline plugin.
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Check out George's blog post _ipcpipeline: Splitting a GStreamer
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pipeline into multiple processes_ or his lightning talk from last
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year's GStreamer Conference in Prague for all the gory details.
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- proxysink and proxysrc are new elements to pass data from one
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pipeline to another within the same process, very similar to the
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existing inter elements, but not limited to raw audio and video
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data. These new proxy elements are very special in how they work
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under the hood, which makes them extremely powerful, but also
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dangerous if not used with care. The reason for this is that it's
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not just data that's passed from sink to src, but these elements
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basically establish a two-way wormhole that passes through queries
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and events in both directions, which means caps negotiation and
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allocation query driven zero-copy can work through this wormhole.
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There are scheduling considerations as well: proxysink forwards
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everything into the proxysrc pipeline directly from the proxysink
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streaming thread. There is a queue element inside proxysrc to
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decouple the source thread from the sink thread, but that queue is
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not unlimited, so it is entirely possible that the proxysink
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pipeline thread gets stuck in the proxysrc pipeline, e.g. when that
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pipeline is paused or stops consuming data for some other reason.
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This means that one should always shut down down the proxysrc
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pipeline before shutting down the proxysink pipeline, for example.
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Or at least take care when shutting down pipelines. Usually this is
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not a problem though, especially not in live pipelines. For more
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information see Nirbheek's blog post _Decoupling GStreamer
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Pipelines_, and also check out out the new ipcpipeline plugin for
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sending data from one process to another process (see above).
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- lcms is a new LCMS-based ICC color profile correction element
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- openmptdec is a new OpenMPT-based decoder for module music formats,
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such as S3M, MOD, XM, IT. It is built on top of a new
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GstNonstreamAudioDecoder base class which aims to unify handling of
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files which do not operate a streaming model. The wildmidi plugin
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has also been revived and is also implemented on top of this new
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base class.
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- The curl plugin has gained a new curlhttpsrc element, which is
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useful for testing HTTP protocol version 2.0 amongst other things.
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- The msdk plugin has gained a MPEG-2 video decoder(msdkmpeg2dec), VP8
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decoder(msdkvp8dec) and a VC1/WMV decoder(msdkvc1dec)
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Noteworthy new API
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- GstPromise provides future/promise-like functionality. This is used
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in the GStreamer WebRTC implementation.
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- GstReferenceTimestampMeta is a new meta that allows you to attach
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additional reference timestamps to a buffer. These timestamps don't
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have to relate to the pipeline clock in any way. Examples of this
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could be an NTP timestamp when the media was captured, a frame
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counter on the capture side or the (local) UNIX timestamp when the
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media was captured. The decklink elements make use of this.
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- GstVideoRegionOfInterestMeta: it's now possible to attach generic
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free-form element-specific parameters to a region of interest meta,
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for example to tell a downstream encoder to use certain codec
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parameters for a certain region.
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- gst_bus_get_pollfd can be used to obtain a file descriptor for the
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bus that can be poll()-ed on for new messages. This is useful for
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integration with non-GLib event loops.
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- gst_get_main_executable_path() can be used by wrapper plugins that
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need to find things in the directory where the application
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executable is located. In the same vein,
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GST_PLUGIN_DEPENDENCY_FLAG_PATHS_ARE_RELATIVE_TO_EXE can be used to
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signal that plugin dependency paths are relative to the main
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executable.
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- pad templates can be told about the GType of the pad subclass of the
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pad via newly-added GstPadTemplate API API or the
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gst_element_class_add_static_pad_template_with_gtype() convenience
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function. gst-inspect-1.0 will use this information to print pad
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properties.
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- new convenience functions to iterate over element pads without using
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the GstIterator API: gst_element_foreach_pad(),
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gst_element_foreach_src_pad(), and gst_element_foreach_sink_pad().
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- GstBaseSrc and appsrc have gained support for buffer lists:
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GstBaseSrc subclasses can use gst_base_src_submit_buffer_list(), and
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applications can use gst_app_src_push_buffer_list() to push a buffer
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list into appsrc.
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- The GstHarness unit test harness has a couple of new convenience
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functions to retrieve all pending data in the harness in form of a
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single chunk of memory.
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- GstAudioStreamAlign is a new helper object for audio elements that
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handles discontinuity detection and sample alignment. It will align
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samples after the previous buffer's samples, but keep track of the
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divergence between buffer timestamps and sample position (jitter).
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If it exceeds a configurable threshold the alignment will be reset.
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This simply factors out code that was duplicated in a number of
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elements into a common helper API.
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- The GstVideoEncoder base class implements Quality of Service (QoS)
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now. This is disabled by default and must be opted in by setting the
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"qos" property, which will make the base class gather statistics
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about the real-time performance of the pipeline from downstream
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elements (usually sinks that sync the pipeline clock). Subclasses
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can then make use of this by checking whether input frames are late
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already using gst_video_encoder_get_max_encode_time() If late, they
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can just drop them and skip encoding in the hope that the pipeline
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will catch up.
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- The GstVideoOverlay interface gained a few helper functions for
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installing and handling a "render-rectangle" property on elements
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that implement this interface, so that this functionality can also
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be used from the command line for testing and debugging purposes.
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The property wasn't added to the interface itself as that would
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require all implementors to provide it which would not be
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backwards-compatible.
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- A new base class, GstNonstreamAudioDecoder for non-stream audio
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decoders was added to gst-plugins-bad. This base-class is meant to
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be used for audio decoders that require the whole stream to be
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loaded first before decoding can start. Examples of this are module
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formats (MOD/S3M/XM/IT/etc), C64 SID tunes, video console music
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files (GYM/VGM/etc), MIDI files and others. The new openmptdec
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element is based on this.
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- Full list of API new in 1.14:
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- GStreamer core API new in 1.14
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- GStreamer base library API new in 1.14
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- gst-plugins-base libraries API new in 1.14
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- gst-plugins-bad: no list, mostly GstWebRTC library and new
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non-stream audio decoder base class.
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New RTP features and improvements
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- rtpulpfecenc and rtpulpfecdec are new elements that implement
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Generic Forward Error Correction (FEC) using Uneven Level Protection
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(ULP) as described in RFC 5109. This can be used to protect against
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certain types of (non-bursty) packet loss, and important packets
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such as those containing codec configuration data or key frames can
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be protected with higher redundancy. Equally, packets that are not
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particularly important can be given low priority or not be protected
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at all. If packets are lost, the receiver can then hopefully restore
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the lost packet(s) from the surrounding packets which were received.
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This is an alternative to, or rather complementary to, dealing with
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packet loss using _retransmission (rtx)_. GStreamer has had
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retransmission support for a long time, but Forward Error Correction
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allows for different trade-offs: The advantage of Forward Error
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Correction is that it doesn't add latency, whereas retransmission
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requires at least one more roundtrip to request and hopefully
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receive lost packets; Forward Error Correction increases the
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required bandwidth however, even in situations where there is no
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packet loss at all, so one will typically want to fine-tune the
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overhead and mechanisms used based on the characteristics of the
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link at the time.
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- New _Redundant Audio Data (RED)_ encoders and decoders for RTP as
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per RFC 2198 are also provided (rtpredenc and rtpreddec), mostly for
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chrome webrtc compatibility, as chrome will wrap ULPFEC-protected
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streams in RED packets, and such streams need to be wrapped and
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unwrapped in order to use ULPFEC with chrome.
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- a few new buffer flags for FEC support:
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GST_BUFFER_FLAG_NON_DROPPABLE can be used to mark important buffers,
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e.g. to flag RTP packets carrying keyframes or codec setup data for
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RTP Forward Error Correction purposes, or to prevent still video
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frames from being dropped by elements due to QoS. There already is a
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GST_BUFFER_FLAG_DROPPABLE. GST_RTP_BUFFER_FLAG_REDUNDANT is used to
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signal internally that a packet represents a redundant RTP packet
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and used in rtpstorage to hold back the packet and use it only for
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recovery from packet loss. Further work is still needed in
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payloaders to make use of these.
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- rtpbin now has an option for increasing timestamp offsets gradually:
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Sudden large changes to the internal ts_offset may cause timestamps
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to move backwards and may also cause visible glitches in media
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playback. The new "max-ts-offset-adjustment" and "max-ts-offset"
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properties let the application control the rate to apply changes to
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ts_offset. There have also been some EOS/BYE handling improvements
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in rtpbin.
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- rtpjitterbuffer has a new fast start mode: in many scenarios the
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jitter buffer will have to wait for the full configured latency
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before it can start outputting packets. The reason for that is that
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it often can't know what the sequence number of the first expected
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RTP packet is, so it can't know whether a packet earlier than the
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earliest packet received will still arrive in future. This behaviour
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can now be bypassed by setting the "faststart-min-packets" property
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to the number of consecutive packets needed to start, and the jitter
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buffer will start output packets as soon as it has N consecutive
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packets queued internally. This is particularly useful to get a
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first video frame decoded and rendered as quickly as possible.
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- rtpL8pay and rtpL8depay provide RTP payloading and depayloading for
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8-bit raw audio
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New element features
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- playbin3 has gained support or gapless playback via the
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"about-to-finish" signal where users can set the uri for the next
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item to play. For non-live streams this will be emitted as soon as
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the first uri has finished downloading, so with sufficiently large
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buffers it is now possible to pre-buffer the next item well ahead of
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time (unlike playbin where there would not be a lot of time between
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"about-to-finish" emission and the end of the stream). If the stream
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format of the next stream is the same as that of the previous
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stream, the data will be concatenated via the concat element.
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Whether this will result in true gaplessness depends on the
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container format and codecs used, there might still be codec-related
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gaps between streams with some codecs.
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- tee now does allocation query aggregation, which is important for
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zero-copy and efficient data handling, especially for video. Those
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who want to drop allocation queries on purpose can use the identity
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element's new "drop-allocation" property for that instead.
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- audioconvert now has a "mix-matrix" property, which obsoletes the
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audiomixmatrix element. There's also mix matrix support in the audio
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conversion and channel mixing API.
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- x264enc: new "insert-vui" property to disable VUI (Video Usability
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Information) parameter insertion into the stream, which allows
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creation of streams that are compatible with certain legacy hardware
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decoders that will refuse to decode in certain combinations of
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resolution and VUI parameters; the max. allowed number of B-frames
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was also increased from 4 to 16.
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- dvdlpcmdec: has gained support for Blu-Ray audio LPCM.
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- appsrc has gained support for buffer lists (see above) and also seen
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some other performance improvements.
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- flvmux has been ported to the GstAggregator base class which means
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it can work in defined-latency mode with live input sources and
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continue streaming if one of the inputs stops producing data.
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- jpegenc has gained a "snapshot" property just like pngenc to make it
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easier to output just a single encoded frame.
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- jpegdec will now handle interlaced MJPEG streams properly and also
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handles frames without an End of Image marker better.
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- v4l2: There are now video encoders for VP8, VP9, MPEG4, and H263.
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The v4l2 video decoder handles dynamic resolution changes, and the
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video4linux device provider now does much faster device probing. The
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plugin also no longer uses the libv4l2 library by default, as it has
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prevented a lot of interesting use cases like CREATE_BUFS, DMABuf,
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usage of TRY_FMT. As the libv4l2 library is totally inactive and not
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really maintained, we decided to disable it. This might affect a
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small number of cheap/old webcams with custom vendor formats for
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which we do not provide conversion in GStreamer. It is possible to
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re-enable support for libv4l2 at run-time however, by setting the
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environment variable GST_V4L2_USE_LIBV4L2=1.
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- rtspsrc now has support for RTSP protocol version 2.0 as well as
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ONVIF audio backchannels (see below for more details). It also
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sports a new "accept-certificate" signal for "manually" checking a
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TLS certificate for validity. It now also prints RTSP/SDP messages
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to the gstreamer debug log instead of stdout.
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- shout2send now uses non-blocking I/O and has a configurable network
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operations timeout.
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- splitmuxsink has gained a "split-now" action signal and new
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"alignment-threshold" and "use-robust-muxing" properties. If robust
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muxing is enabled, it will check and set the muxer's reserved space
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properties if present. This is primarily for use with mp4mux's
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robust muxing mode.
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- qtmux has a new _prefill recording mode_ which sets up a moov header
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with the correct sample positions beforehand, which then allows
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software like Adobe Premiere and FinalCut Pro to import the files
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while they are still being written to. This only works with constant
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framerate I-frame only streams, and for now only support for ProRes
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video and raw audio is implemented. Adding support for additional
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codecs is just a matter of defining appropriate maximum frame sizes
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though.
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- qtmux also supports writing of svmi atoms with stereoscopic video
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information now. Trak timescales can be configured on a per-stream
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basis using the "trak-timescale" property on the sink pads. Various
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new formats can be muxed: MPEG layer 1 and 2, AC3 and Opus, as well
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as PNG and VP9.
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- souphttpsrc now does connection sharing by default: it shares its
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SoupSession with other elements in the same pipeline via a
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GstContext if possible (session-wide settings are all the defaults).
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This allows for connection reuse, cookie sharing, etc. Applications
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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
|
|
|
|
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.
|
|
|
|
Plugin removals
|
|
|
|
- the sunaudio plugin was removed, since it couldn't ever have been
|
|
built or used with GStreamer 1.0, but no one even noticed in all
|
|
these years.
|
|
|
|
- the 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
|
|
|
|
- some MPEG video parser API in the API unstable codecutils library in
|
|
gst-plugins-bad was removed after having been deprecated for 5
|
|
years.
|
|
|
|
|
|
Miscellaneous changes
|
|
|
|
- The video support library has gained support for a few new pixel
|
|
formats:
|
|
- NV16_10LE32: 10-bit variant of NV16, packed into 32bit words (plus 2
|
|
bits padding)
|
|
- NV12_10LE32: 10-bit variant of NV12, packed into 32bit words (plus 2
|
|
bits padding)
|
|
- GRAY10_LE32: 10-bit grayscale, packed in 32bit words (plus 2 bits
|
|
padding)
|
|
|
|
- decodebin, playbin and GstDiscoverer have seen stability
|
|
improvements in corner cases such as shutdown while still starting
|
|
up or shutdown in error cases (hat tip to the oss-fuzz project).
|
|
|
|
- floating reference handling was inconsistent and has been cleaned up
|
|
across the board, including annotations. This solves various
|
|
long-standing memory leaks in language bindings, which e.g. often
|
|
caused elements and pads to be leaked.
|
|
|
|
- major gobject-introspection annotation improvements for large parts
|
|
of the library API, including nullability of return types and
|
|
function parameters, correct types (e.g. strings vs. filenames),
|
|
ownership transfer, array length parameters, etc. This allows to use
|
|
bigger parts of the GStreamer API to be safely used from dynamic
|
|
language bindings (e.g. Python, Javascript) and allows static
|
|
bindings (e.g. C#, Rust, Vala) to autogenerate more API bindings
|
|
without manual intervention.
|
|
|
|
OpenGL integration
|
|
|
|
- The GStreamer OpenGL integration library has moved to
|
|
gst-plugins-base and is now part of our stable API.
|
|
|
|
- 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.
|
|
|
|
- Prefer wayland display over X11: As most Wayland compositors support
|
|
XWayland, the X11 backend would get selected.
|
|
|
|
- gldownload can export dmabufs now, and glupload will advertise
|
|
dmabuf as caps feature.
|
|
|
|
|
|
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.
|
|
|
|
|
|
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
|
|
|
|
|
|
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.
|
|
|
|
|
|
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.
|
|
|
|
|
|
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
|
|
|
|
|
|
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
|
|
|
|
- Removed the need for "glue" code
|
|
|
|
- Provide a nuget
|
|
|
|
- Misc API fixes
|
|
|
|
|
|
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.
|
|
|
|
|
|
Platform-specific improvements
|
|
|
|
Android
|
|
|
|
- ahcsrc (Android camera source) does autofocus now
|
|
|
|
macOS and iOS
|
|
|
|
- this section will be filled in shortly {FIXME!}
|
|
|
|
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.
|
|
|
|
|
|
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,
|
|
|
|
... and many others who have contributed bug reports, translations, sent
|
|
suggestions or helped testing.
|
|
|
|
|
|
Bugs fixed in 1.14
|
|
|
|
More than 800 bugs have been fixed during the development of 1.14.
|
|
|
|
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.
|
|
|
|
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
|
|
|
|
After the 1.14.0 release there will be several 1.14.x bug-fix releases
|
|
which will contain bug fixes which have been deemed suitable for a
|
|
stable branch, but no new features or intrusive changes will be added to
|
|
a bug-fix release usually. The 1.14.x bug-fix releases will be made from
|
|
the git 1.14 branch, which is a stable branch.
|
|
|
|
1.14.0
|
|
|
|
1.14.0 was released on 19 March 2018.
|
|
|
|
1.14.1
|
|
|
|
The first 1.14 bug-fix release (1.14.1) is scheduled to be released
|
|
around the end of March or beginning of April.
|
|
|
|
This release only contains bugfixes and it should be safe to update from
|
|
1.14.0.
|
|
|
|
|
|
Known Issues
|
|
|
|
- The webrtcdsp element (which is unrelated to the newly-landed
|
|
GStreamer webrtc support) is currently not shipped as part of the
|
|
Windows binary packages due to a build system issue.
|
|
|
|
|
|
Schedule for 1.16
|
|
|
|
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
|
|
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,
|
|
1.6, 1.4, 1.2 and 1.0 release series.
|
|
|
|
------------------------------------------------------------------------
|
|
|
|
_These release notes have been prepared by Tim-Philipp Müller with_
|
|
_contributions from Sebastian Dröge, Sreerenj Balachandran, Thibault
|
|
Saunier_ _and Víctor Manuel Jáquez Leal._
|
|
|
|
_License: CC BY-SA 4.0_
|