-# GStreamer 1.8 Release Notes
-**GStreamer 1.8.0 was released on 24 March 2016.**
-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 1.14 RELEASE NOTES
-As always, this release is again packed with new features, bug fixes and other
-improvements.
-See [https://gstreamer.freedesktop.org/releases/1.8/][latest] for the latest
-version of this document.
-
-*Last updated: Thursday 24 March 2016, 10:00 UTC [(log)][gitlog]*
-
-[latest]: https://gstreamer.freedesktop.org/releases/1.8/
-[gitlog]: https://cgit.freedesktop.org/gstreamer/www/log/src/htdocs/releases/1.8/release-notes-1.8.md
-
-## Highlights
-
-- **Hardware-accelerated zero-copy video decoding on Android**
-
-- **New video capture source for Android using the android.hardware.Camera API**
-
-- **Windows Media reverse playback** support (ASF/WMV/WMA)
-
-- **New tracing system** provides support for more sophisticated debugging tools
-
-- **New high-level GstPlayer playback convenience API**
-
-- **Initial support for the new [Vulkan][vulkan] API**, see
- [Matthew Waters' blog post][vulkan-in-gstreamer] for more details
-
-- **Improved Opus audio codec support**: Support for more than two channels; MPEG-TS demuxer/muxer can now handle Opus;
- [sample-accurate][opus-sample-accurate] encoding/decoding/transmuxing with
- Ogg, Matroska, ISOBMFF (Quicktime/MP4), and MPEG-TS as container;
- [new codec utility functions for Opus header and caps handling][opus-codec-utils]
- in pbutils library. The Opus encoder/decoder elements were also moved to
- gst-plugins-base (from -bad), and the opus RTP depayloader/payloader to -good.
-
- [opus-sample-accurate]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstaudiometa.html#GstAudioClippingMeta
- [opus-codec-utils]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstpbutilscodecutils.html
-
-- **GStreamer VAAPI module now released and maintained as part of the GStreamer project**
-
- [vulkan]: https://www.khronos.org/vulkan
- [vulkan-in-gstreamer]: http://ystreet00.blogspot.co.uk/2016/02/vulkan-in-gstreamer.html
-
-## Major new features and changes
-
-### Noteworthy new API, features and other changes
-
-- New GstVideoAffineTransformationMeta meta for adding a simple 4x4 affine
- transformation matrix to video buffers
-
-- [g\_autoptr()](https://developer.gnome.org/glib/stable/glib-Miscellaneous-Macros.html#g-autoptr)
- support for all types is exposed in GStreamer headers now, in combination
- with a sufficiently-new GLib version (i.e. 2.44 or later). This is primarily
- for the benefit of application developers who would like to make use of
- this, the GStreamer codebase itself will not be using g_autoptr() for
- the time being due to portability issues.
-
-- GstContexts are now automatically propagated to elements added to a bin
- or pipeline, and elements now maintain a list of contexts set on them.
- The list of contexts set on an element can now be queried using the new functions
- [gst\_element\_get\_context()](https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstElement.html#gst-element-get-context)
- and [gst\_element\_get\_contexts()](https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstElement.html#gst-element-get-contexts). GstContexts are used to share context-specific configuration objects
- between elements and can also be used by applications to set context-specific
- configuration objects on elements, e.g. for OpenGL or Hardware-accelerated
- video decoding.
-
-- New [GST\_BUFFER\_DTS\_OR\_PTS()](https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstBuffer.html#GST-BUFFER-DTS-OR-PTS:CAPS)
- convenience macro that returns the decode timestamp if one is set and
- otherwise returns the presentation timestamp
-
-- New GstPadEventFullFunc that returns a GstFlowReturn instead of a gboolean.
- This new API is mostly for internal use and was added to fix a race condition
- where occasionally internal flow error messages were posted on the bus when
- sticky events were propagated at just the wrong moment whilst the pipeline
- was shutting down. This happened primarily when the pipeline was shut down
- immediately after starting it up. GStreamer would not know that the reason
- the events could not be propagated was because the pipeline was shutting down
- and not some other problem, and now the flow error allows GStreamer to know
- the reason for the failure (and that there's no reason to post an error
- message). This is particularly useful for queue-like elements which may need
- to asynchronously propagate a previous flow return from downstream.
-
-- Pipeline dumps in form of "dot files" now also show pad properties that
- differ from their default value, the same as it does for elements. This is
- useful for elements with pad subclasses that provide additional properties,
- e.g. videomixer or compositor.
-
-- Pad probes are now guaranteed to be called in the order they were added
- (before they were called in reverse order, but no particular order was
- documented or guaranteed)
-
-- Plugins can now have dependencies on device nodes (not just regular files)
- and also have a prefix filter. This is useful for plugins that expose
- features (elements) based on available devices, such as the video4linux
- plugin does with video decoders on certain embedded systems.
-
-- gst\_segment\_to\_position() has been deprecated and been replaced by the
- better-named gst\_segment\_position\_from\_running\_time(). At the same time
- gst\_segment\_position\_from\_stream\_time() was added, as well as \_full()
- variants of both to deal with negative stream time.
-
-- GstController: the interpolation control source gained a new monotonic cubic
- interpolation mode that, unlike the existing cubic mode, will never overshoot
- the min/max y values set.
-
-- GstNetAddressMeta: can now be read from buffers in language bindings as well,
- via the new gst\_buffer\_get\_net\_address\_meta() function
-
-- ID3 tag PRIV frames are now extraced into a new GST\_TAG\_PRIVATE\_DATA tag
-
-- gst-launch-1.0 and gst\_parse\_launch() now warn in the most common case if
- a dynamic pad link could not be resolved, instead of just silently
- waiting to see if a suitable pad appears later, which is often perceived
- by users as hanging -- they are now notified when this happens and can check
- their pipeline.
-
-- GstRTSPConnection now also parses custom RTSP message headers and retains
- them for the application instead of just ignoring them
-
-- rtspsrc handling of authentication over tunneled connections (e.g. RTSP over HTTP)
- was fixed
-
-- gst\_video\_convert\_sample() now crops if there is a crop meta on the input buffer
-
-- The debugging system printf functions are now exposed for general use, which
- supports special printf format specifiers such as GST\_PTR\_FORMAT and
- GST\_SEGMENT\_FORMAT to print GStreamer-related objects. This is handy for
- systems that want to prepare some debug log information to be output at a
- later point in time. The GStreamer-OpenGL subsystem is making use of these
- new functions, which are [gst\_info\_vasprintf()][gst_info_vasprintf],
- [gst\_info\_strdup\_vprintf()][gst_info_strdup_vprintf] and
- [gst\_info\_strdup\_printf()][gst_info_strdup_printf].
-
-- videoparse: "strides", "offsets" and "framesize" properties have been added to
- allow parsing raw data with strides and padding that do not match GStreamer
- defaults.
-
-- GstPreset reads presets from the directories given in GST\_PRESET\_PATH now.
- Presets are read from there after presets in the system path, but before
- application and user paths.
-
-[gst_info_vasprintf]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/gstreamer-GstInfo.html#gst-info-vasprintf
-[gst_info_strdup_vprintf]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/gstreamer-GstInfo.html#gst-info-strdup-vprintf
-[gst_info_strdup_printf]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/gstreamer-GstInfo.html#gst-info-strdup-printf
-
-### New Elements
-
-- [netsim](): a new (resurrected) element to simulate network jitter and
- packet dropping / duplication.
-
-- New VP9 RTP payloader/depayloader elements: rtpvp9pay/rtpvp9depay
-
-- New [videoframe_audiolevel]() element, a video frame synchronized audio level element
-
-- New spandsp-based tone generator source
-
-- New NVIDIA NVENC-based H.264 encoder for GPU-accelerated video encoding on
- suitable NVIDIA hardware
-
-- [rtspclientsink](), a new RTSP RECORD sink element, was added to gst-rtsp-server
-
-- [alsamidisrc](), a new ALSA MIDI sequencer source element
-
-### Noteworthy element features and additions
-
-- *identity*: new ["drop-buffer-flags"](https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer-plugins/html/gstreamer-plugins-identity.html#GstIdentity--drop-buffer-flags)
- property to drop buffers based on buffer flags. This can be used to drop all
- non-keyframe buffers, for example.
-
-- *multiqueue*: various fixes and improvements, in particular special handling
- for sparse streams such as substitle streams, to make sure we don't overread
- them any more. For sparse streams it can be normal that there's no buffer for
- a long period of time, so having no buffer queued is perfectly normal. Before
- we would often unnecessarily try to fill the subtitle stream queue, which
- could lead to much more data being queued in multiqueue than necessary.
-
-- *multiqueue*/*queue*: When dealing with time limits, these elements now use the
- new ["GST_BUFFER_DTS_OR_PTS"](https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstBuffer.html#GST-BUFFER-DTS-OR-PTS:CAPS)
- and ["gst_segment_to_running_time_full()"](https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstSegment.html#gst-segment-to-running-time-full)
- API, resulting in more accurate levels, especially when dealing with non-raw
- streams (where reordering happens, and we want to use the increasing DTS as
- opposed to the non-continuously increasing PTS) and out-of-segment input/output.
- Previously all encoded buffers before the segment start, which can happen when
- doing ACCURATE seeks, were not taken into account in the queue level calculation.
-
-- *multiqueue*: New ["use-interleave"](https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer-plugins/html/gstreamer-plugins-multiqueue.html#GstMultiQueue--use-interleave)
- property which allows the size of the queues to be optimized based on the input
- streams interleave. This should only be used with input streams which are properly
- timestamped. It will be used in the future decodebin3 element.
-
-- *queue2*: new ["avg-in-rate"](https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer-plugins/html/gstreamer-plugins-queue2.html#GstQueue2--avg-in-rate)
- property that returns the average input rate in bytes per second
-
-- audiotestsrc now supports all audio formats and is no longer artificially
- limited with regard to the number of channels or sample rate
-
-- gst-libav (ffmpeg codec wrapper): map and enable JPEG2000 decoder
-
-- multisocketsink can, on request, send a custom GstNetworkMessage event
- upstream whenever data is received from a client on a socket. Similarly,
- socketsrc will, on request, pick up GstNetworkMessage events from downstream
- and send any data contained within them via the socket. This allows for
- simple bidirectional communication.
-
-- matroska muxer and demuxer now support the ProRes video format
-
-- Improved VP8/VP9 decoding performance on multi-core systems by enabling
- multi-threaded decoding in the libvpx-based decoders on such systems
-
-- appsink has a new ["wait-on-eos"](https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-plugins/html/gst-plugins-base-plugins-appsink.html#GstAppSink--wait-on-eos)
- property, so in cases where it is uncertain if an appsink will have a consumer for
- its buffers when it receives an EOS this can be set to FALSE to ensure that the
- appsink will not hang.
-
-- rtph264pay and rtph265pay have a new "config-interval" mode -1 that will
- re-send the setup data (SPS/PPS/VPS) before every keyframe to ensure
- optimal coverage and the shortest possibly start-up time for a new client
-
-- mpegtsmux can now mux H.265/HEVC video as well
-
-- The MXF muxer was ported to 1.x and produces more standard conformant files now
- that can be handled by more other software; The MXF demuxer got improved
- support for seek tables (IndexTableSegments).
-
-### Plugin moves
-
-- The rtph265pay/depay RTP payloader/depayloader elements for H.265/HEVC video
- from the rtph265 plugin in -bad have been moved into the existing rtp plugin
- in gst-plugins-good.
-
-- The mpg123 plugin containing a libmpg123 based audio decoder element has
- been moved from -bad to -ugly.
-
-- The Opus encoder/decoder elements have been moved to gst-plugins-base and
- the RTP payloader to gst-plugins-good, both coming from gst-plugins-bad.
+GStreamer 1.14.0 has not been released yet. It is scheduled for release
+in early March 2018.
-### New tracing tools for developers
+There are unstable pre-releases available for testing and development
+purposes. The latest pre-release is version 1.13.91 (rc2) and was
+released on 12 March 2018.
-A new tracing subsystem API has been added to GStreamer, which provides
-external tracers with the possibility to strategically hook into GStreamer
-internals and collect data that can be evaluated later. These tracers are a
-new type of plugin features, and GStreamer core ships with a few example
-tracers (latency, stats, rusage, log) to start with. Tracers can be loaded
-and configured at start-up via an environment variable (GST\_TRACER\_PLUGINS).
-
-Background: While GStreamer provides plenty of data on what's going on in a
-pipeline via its debug log, that data is not necessarily structured enough to
-be generally useful, and the overhead to enable logging output for all data
-required might be too high in many cases. The new tracing system allows tracers
-to just obtain the data needed at the right spot with as little overhead as
-possible, which will be particularly useful on embedded systems.
-
-Of course it has always been possible to do performance benchmarks and debug
-memory leaks, memory consumption and invalid memory access using standard
-operating system tools, but there are some things that are difficult to track
-with the standard tools, and the new tracing system helps with that. Examples
-are things such as latency handling, buffer flow, ownership transfer of
-events and buffers from element to element, caps negotiation, etc.
-
-For some background on the new tracing system, watch Stefan Sauer's
-GStreamer Conference talk ["A new tracing subsystem for GStreamer"][tracer-0]
-and for a more specific example how it can be useful have a look at
-Thiago Santos's lightning talk ["Analyzing caps negotiation using GstTracer"][tracer-1]
-and his ["GstTracer experiments"][tracer-2] blog post. There was also a Google
-Summer of Code project in 2015 that used tracing system for a graphical
-GStreamer debugging tool ["gst-debugger"][tracer-3].
-
-This is all still very much work in progress, but we hope this will provide the
-foundation for a whole suite of new debugging tools for GStreamer pipelines.
-
-[tracer-0]: https://gstconf.ubicast.tv/videos/a-new-tracing-subsystem-for-gstreamer/
-[tracer-1]: https://gstconf.ubicast.tv/videos/analyzing-caps-negotiation-using-gsttracer/
-[tracer-2]: http://blog.thiagoss.com/2015/07/23/gsttracer-experiments/
-[tracer-3]: https://git.gnome.org/browse/gst-debugger
-
-### GstPlayer: a new high-level API for cross-platform multimedia playback
-
-GStreamer has had reasonably high-level API for multimedia playback
-in the form of the playbin element for a long time. This allowed application
-developers to just configure a URI to play, and playbin would take care of
-everything else. This works well, but there is still way too much to do on
-the application-side to implement a fully-featured playback application, and
-too much general GStreamer pipeline API exposed, making it less accessible
-to application developers.
-
-Enter GstPlayer. GstPlayer's aim is to provide an even higher-level abstraction
-of a fully-featured playback API but specialised for its specific use case. It
-also provides easy integration with and examples for Gtk+, Qt, Android, OS/X,
-iOS and Windows. Watch Sebastian's [GstPlayer talk at the GStreamer Conference][gstplayer-talk]
-for more information, or check out the [GstPlayer API reference][gstplayer-api]
-and [GstPlayer examples][gstplayer-examples].
-
-[gstplayer-api]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-bad-libs/html/player.html
-[gstplayer-talk]: https://gstconf.ubicast.tv/videos/gstplayer-a-simple-cross-platform-api-for-all-your-media-playback-needs-part-1/
-[gstplayer-examples]: https://github.com/sdroege/gst-player/
-
-### Adaptive streaming: DASH, HLS and MSS improvements
-
-- dashdemux now supports loading external xml nodes pointed from its MPD.
-
-- Content protection nodes parsing support for PlayReady WRM in mssdemux.
-
-- Reverse playback was improved to respect seek start and stop positions.
-
-- Adaptive demuxers (hlsdemux, dashdemux, mssdemux) now support the SNAP_AFTER
- and SNAP_BEFORE seek flags which will jump to the nearest fragment boundary
- when executing a seek, which means playback resumes more quickly after a seek.
-
-### Audio library improvements
-
-- audio conversion, quantization and channel up/downmixing functionality
- has been moved from the audioconvert element into the audio library and
- is now available as public API in form of [GstAudioConverter][audio-0],
- [GstAudioQuantize][audio-1] and [GstAudioChannelMixer][audio-2].
- Audio resampling will follow in future releases.
-
-- [gst\_audio\_channel\_get\_fallback\_mask()][audio-3] can be used
- to retrieve a default channel mask for a given number of channels as last
- resort if the layout is unknown
-
-- A new [GstAudioClippingMeta][audio-4] meta was added for specifying clipping
- on encoded audio buffers
-
-- A new GstAudioVisualizer base class for audio visualisation elements;
- most of the existing visualisers have been ported over to the new base class.
- This new base class lives in the pbutils library rather than the audio library,
- since we'd have had to make libgstaudio depend on libgstvideo otherwise,
- which was deemed undesirable.
-
-[audio-0]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-GstAudioConverter.html
-[audio-1]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-GstAudioQuantize.html
-[audio-2]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstaudiochannels.html#gst-audio-channel-mix-new
-[audio-3]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstaudiochannels.html#gst-audio-channel-get-fallback-mask
-[audio-4]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstaudiometa.html#GstAudioClippingMeta
-
-### GStreamer OpenGL support improvements
-
-#### Better OpenGL Shader support
-
-[GstGLShader][shader] has been revamped to allow more OpenGL shader types
-by utilizing a new GstGLSLStage object. Each stage holds an OpenGL pipeline
-stage such as a vertex, fragment or a geometry shader that are all compiled
-separately into a program that is executed.
-
-The glshader element has also received a revamp as a result of the changes in
-the library. It does not take file locations for the vertex and fragment
-shaders anymore. Instead it takes the strings directly leaving the file
-management to the application.
-
-A new [example][liveshader-example] was added utilizing the new shader
-infrastructure showcasing live shader edits.
-
-[shader]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-bad-libs/html/gst-plugins-bad-libs-gstglshader.html
-[liveshader-example]: https://cgit.freedesktop.org/gstreamer/gst-plugins-bad/tree/tests/examples/gtk/glliveshader.c
-
-#### OpenGL GLMemory rework
-
-[GstGLMemory] was extensively reworked to support the addition of multiple
-texture targets required for zero-copy integration with the Android
-MediaCodec elements. This work was also used to provide IOSurface based
-GLMemory on OS X for zero-copy with OS X's VideoToolbox decoder (vtdec) and
-AV Foundation video source (avfvideosrc). There are also patches in bugzilla
-for GstGLMemoryEGL specifically aimed at improving the decoding performance on
-the Raspberry Pi.
-
-[GstGLMemory]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-bad-libs/html/gst-plugins-bad-libs-gstglmemory.html
-
-A texture-target field was added to video/x-raw(memory:GLMemory) caps to signal
-the texture target contained in the GLMemory. Its values can be 2D, rectangle
-or external-oes. glcolorconvert can convert between the different formats as
-required and different elements will accept or produce different targets. e.g.
-glimagesink can take and render external-oes textures directly as required for
-effecient zero-copy on android.
-
-A generic GL allocation framework was also implemented to support the generic
-allocation of OpenGL buffers and textures which is used extensively by
-GstGLBufferPool.
-
-#### OpenGL DMABuf import uploader
-
-There is now a DMABuf uploader available for automatic selection that will
-attempt to import the upstream provided DMABuf. The uploader will import into
-2D textures with the necesarry format. YUV to RGB conversion is still provided
-by glcolorconvert to avoid the laxer restrictions with external-oes textures.
-
-#### OpenGL queries
-
-Queries of various aspects of the OpenGL runtime such as timers, number of
-samples or the current timestamp are not possible. The GstGLQuery object uses a
-delayed debug system to delay the debug output to later to avoid expensive calls
-to the glGet\* family of functions directly after finishing a query. It is
-currently used to output the time taken to perform various operations of texture
-uploads and downloads in GstGLMemory.
-
-#### New OpenGL elements
-
-glcolorbalance has been created mirroring the videobalance elements.
-glcolorbalance provides the exact same interface as videobalance so can be used
-as a GPU accelerated replacement. glcolorbalance has been added to glsinkbin so
-usage with playsink/playbin will use it automatically instead of videobalance
-where possible.
-
-glvideoflip, which is the OpenGL equiavalant of videoflip, implements the exact
-same interface and functionality as videoflip.
-
-#### EGL implementation now selects OpenGL 3.x
-
-The EGL implementation can now select OpenGL 3.x contexts.
-
-#### OpenGL API removal
-
-The GstGLDownload library object was removed as it was not used by anything.
-Everything is performed by GstGLMemory or in the gldownloadelement.
-
-The GstGLUploadMeta library object was removed as it was not being used and we
-don't want to promote the use of GstVideoGLTextureUploadMeta.
-
-#### OpenGL: Other miscellaneous changes
-
-- The EGL implementation can now select OpenGL 3.x contexts. This brings
- OpenGL 3.x to e.g. wayland and other EGL systems.
-
-- glstereomix/glstereosplit are now built and are usable on OpenGL ES systems
-
-- The UYVY/YUY2 to RGBA and RGBA to UYVY/YUY2 shaders were fixed removing the
- sawtooth pattern and luma bleeding.
-
-- We now utilize the GL\_APPLE\_sync extension on iOS devices which improves
- performance of OpenGL applications, especially with multiple OpenGL
- contexts.
-
-- glcolorconvert now uses a bufferpool to avoid costly
- glGenTextures/glDeleteTextures for every frame.
-
-- glvideomixer now has full glBlendFunc and glBlendEquation support per input.
-
-- gltransformation now support navigation events so your weird transformations
- also work with DVD menus.
-
-- qmlglsink can now run on iOS, OS X and Android in addition to the already
- supported Linux platform.
-
-- glimagesink now posts unhandled keyboard and mouse events (on backends that
- support user input, current only X11) on the bus for the application.
-
-### Initial GStreamer Vulkan support
-
-Some new elements, vulkansink and vulkanupload have been implemented utilizing
-the new Vulkan API. The implementation is currently limited to X11 platforms
-(via xcb) and does not perform any scaling of the stream's contents to the size
-of the available output.
+See https://gstreamer.freedesktop.org/releases/1.14/ for the latest
+version of this document.
-A lot of infrasctructure work has been undertaken to support using Vulkan in
-GStreamer in the future. A number of GstMemory subclasses have been created for
-integrating Vulkan's GPU memory handling along with VkBuffer's and VkImage's
-that can be passed between elements. Some GStreamer refcounted wrappers for
-global objects such as VkInstance, VkDevice, VkQueue, etc have also been
-implemented along with GstContext integration for sharing these objects with the
-application.
+_Last updated: Monday 12 March 2018, 18:00 UTC (log)_
-### GStreamer VAAPI support for hardware-accelerated video decoding and encoding on Intel (and other) platforms
-#### GStreamer VAAPI is now part of upstream GStreamer
+Introduction
-The GStreamer-VAAPI module which provides support for hardware-accelerated
-video decoding, encoding and post-processing on Intel graphics hardware
-on Linux has moved from its previous home at the [Intel Open Source Technology Center][iostc]
-to the upstream GStreamer repositories, where it will in future be maintained
-as part of the upstream GStreamer project and released in lockstep with the
-other GStreamer modules. The current maintainers will continue to spearhead
-the development at the new location:
+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!
-[http://cgit.freedesktop.org/gstreamer/gstreamer-vaapi/][gst-vaapi-git]
+As always, this release is again packed with new features, bug fixes and
+other improvements.
-[gst-vaapi-git]: http://cgit.freedesktop.org/gstreamer/gstreamer-vaapi/
-GStreamer-VAAPI relies heavily on certain GStreamer infrastructure API that
-is still in flux such as the OpenGL integration API or the codec parser
-libraries, and one of the goals of the move was to be able to leverage
-new developments early and provide tighter integration with the latest
-developments of those APIs and other graphics-related APIs provided by
-GStreamer, which should hopefully improve performance even further and in
-some cases might also provide better stability.
+Highlights
-Thanks to everyone involved in making this move happen!
+- WebRTC support: real-time audio/video streaming to and from web
+ browsers
-#### GStreamer VAAPI: Bug tracking
+- Experimental support for the next-gen royalty-free AV1 video codec
-Bugs had already been tracked on [GNOME bugzilla](bgo) but will be moved
-from the gstreamer-vaapi product into a new gstreamer-vaapi component of
-the GStreamer product in bugzilla. Please file new bugs against the new
-component in the GStreamer product from now on.
+- Video4Linux: encoding support, stable element names and faster
+ device probing
-#### GStreamer VAAPI: Pending patches
+- Support for the Secure Reliable Transport (SRT) video streaming
+ protocol
-The code base has been re-indented to the GStreamer code style, which
-affected some files more than others. This means that some of the patches
-in bugzilla might not apply any longer, so if you have any unmerged patches
-sitting in bugzilla please consider checking if they still apply cleany and
-refresh them if not. Sorry for any inconvenience this may cause.
+- RTP Forward Error Correction (FEC) support (ULPFEC)
-#### GStreamer VAAPI: New versioning scheme and supported GStreamer versions
+- RTSP 2.0 support in rtspsrc and gst-rtsp-server
-The version numbering has been changed to match the GStreamer version
-numbering to avoid confusion: there is a new gstreamer-vaapi 1.6.0 release
-and a 1.6 branch that is roughly equivalent to the previous 0.7.0 version.
-Future releases 1.7.x and 1.8.x will be made alongside GStreamer releases.
+- ONVIF audio backchannel support in gst-rtsp-server and rtspsrc
-While it was possible and supported by previous releases to build against
-a whole range of different GStreamer versions (such as 1.2, 1.4, 1.6 or 1.7/1.8),
-in the future there will only be one target branch, so that git master will
-track GStreamer git master, 1.8.x will target GStreamer 1.8, and
-1.6.x will target the 1.6 series.
+- playbin3 gapless playback and pre-buffering support
-[iostc]: http://01.org
-[bgo]: http://bugzilla.gnome.og
+- tee, our stream splitter/duplication element, now does allocation
+ query aggregation which is important for efficient data handling and
+ zero-copy
-#### GStreamer VAAPI: Miscellaneous changes
+- 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.
-All GStreamer-VAAPI functionality is now provided solely by its GStreamer
-elements. There is no more public library exposing GstVaapi API, this API
-was only ever meant for private use by the elements. Parts of it may be
-resurrected again in future if needed, but for now it has all been made
-private.
+- rtpjitterbuffer fast-start mode and timestamp offset adjustment
+ smoothing
-GStreamer-VAAPI now unconditionally uses the codecparser library in
-gst-plugins-bad instead of shipping its own internal copy. Similarly,
-it no longer ships its own codec parsers but relies on the upstream
-codec parser elements.
+- souphttpsrc connection sharing, which allows for connection reuse,
+ cookie sharing, etc.
-The GStreamer-VAAPI encoder elements have been renamed from vaapiencode_foo
-to vaapifooenc, so encoders are now called vaapih264enc, vaapih265enc,
-vaapimpeg2enc, vaapijpegenc, and vaapivp8enc. With this change we now follow
-the standard names in GStreamer, and the plugin documentation is generated
-correctly.
+- nvdec: new plugin for hardware-accelerated video decoding using the
+ NVIDIA NVDEC API
-In the case of the decoders, only the jpeg decoder has been split from the
-general decoding element vaapidecode: vaapijpegdec. This is the first step to
-split per codec each decoding element. The vaapijpegdec has also been given
-marginal rank for the time being.
+- Adaptive DASH trick play support
-#### GStreamer VAAPI: New features in 1.8: 10-bit H.265/HEVC decoding support
+- ipcpipeline: new plugin that allows splitting a pipeline across
+ multiple processes
-Support for decoding 10-bit H.265/HEVC has been added. For the time being
-this only works in combination with vaapisink though, until support for the
-P010 video format used internally is added to GStreamer and to the
-vaGetImage()/vaPutimage() API in the vaapi-intel-driver.
+- Major gobject-introspection annotation improvements for large parts
+ of the library API
-Several fixes for memory leaks, build errors, and in the internal
-video parsing.
-Finally, vaapisink now posts the unhandled keyboard and mouse events to the
-application.
+Major new features and changes
-### GStreamer Video 4 Linux Support
+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.
-Colorimetry support has been enhanced even more. It will now properly select
-default values when not specified by the driver. The range of color formats
-supported by GStreamer has been greatly improved. Notably, support for
-multi-planar I420 has been added along with all the new and non-ambiguous RGB
-formats that got added in recent kernels.
+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.
-The device provider now exposes a variety of properties as found in the udev
-database.
+For more details, background and example code, check out Nirbheek's blog
+post _GStreamer has grown a WebRTC implementation_, as well as Matthew's
+_GStreamer WebRTC_ talk from last year's GStreamer Conference in Prague.
+
+New Elements
+
+- webrtcbin handles the transport aspects of webrtc connections (see
+ WebRTC section above for more details)
+
+- New srtsink and srtsrc elements for the Secure Reliable Transport
+ (SRT) video streaming protocol, which aims to be easy to use whilst
+ striking a new balance between reliability and latency for low
+ latency video streaming use cases. More details about SRT and the
+ implementation in GStreamer in Olivier's blog post _SRT in
+ GStreamer_.
+
+- av1enc and av1dec elements providing experimental support for the
+ next-generation royalty free video AV1 codec, alongside Matroska
+ support for it.
+
+- hlssink2 is a rewrite of the existing hlssink element, but unlike
+ its predecessor hlssink2 takes elementary streams as input and
+ handles the muxing to MPEG-TS internally. It also leverages
+ splitmuxsink internally to do the splitting. This allows more
+ control over the chunk splitting and sizing process and relies less
+ on the co-operation of an upstream muxer. Different to the old
+ hlssink it also works with pre-encoded streams and does not require
+ close interaction with an upstream encoder element.
+
+- audiolatency is a new element for measuring audio latency end-to-end
+ and is useful to measure roundtrip latency including both the
+ GStreamer-internal latency as well as latency added by external
+ components or circuits.
+
+- 'fakevideosink is basically a null sink for video data and very
+ similar to fakesink, only that it will answer allocation queries and
+ will advertise support for various video-specific things such
+ GstVideoMeta, GstVideoCropMeta and GstVideoOverlayCompositionMeta
+ like a normal video sink would. This is useful for throughput
+ testing and testing the zero-copy path when creating a new pipeline.
+
+- ipcpipeline: new plugin that allows the splitting of a pipeline into
+ multiple processes. Usually a GStreamer pipeline runs in a single
+ process and parallelism is achieved by distributing workloads using
+ multiple threads. This means that all elements in the pipeline have
+ access to all the other elements' memory space however, including
+ that of any libraries used. For security reasons one might therefore
+ want to put sensitive parts of a pipeline such as DRM and decryption
+ handling into a separate process to isolate it from the rest of the
+ pipeline. This can now be achieved with the new ipcpipeline plugin.
+ Check out George's blog post _ipcpipeline: Splitting a GStreamer
+ pipeline into multiple processes_ or his lightning talk from last
+ year's GStreamer Conference in Prague for all the gory details.
+
+
+- proxysink and proxysrc are new elements to pass data from one
+ pipeline to another within the same process, very similar to the
+ existing inter elements, but not limited to raw audio and video
+ data. These new proxy elements are very special in how they work
+ under the hood, which makes them extremely powerful, but also
+ dangerous if not used with care. The reason for this is that it's
+ not just data that's passed from sink to src, but these elements
+ basically establish a two-way wormhole that passes through queries
+ and events in both directions, which means caps negotiation and
+ allocation query driven zero-copy can work through this wormhole.
+ There are scheduling considerations as well: proxysink forwards
+ everything into the proxysrc pipeline directly from the proxysink
+ streaming thread. There is a queue element inside proxysrc to
+ decouple the source thread from the sink thread, but that queue is
+ not unlimited, so it is entirely possible that the proxysink
+ pipeline thread gets stuck in the proxysrc pipeline, e.g. when that
+ pipeline is paused or stops consuming data for some other reason.
+ This means that one should always shut down down the proxysrc
+ pipeline before shutting down the proxysink pipeline, for example.
+ Or at least take care when shutting down pipelines. Usually this is
+ not a problem though, especially not in live pipelines. For more
+ information see Nirbheek's blog post _Decoupling GStreamer
+ Pipelines_, and also check out out the new ipcpipeline plugin for
+ sending data from one process to another process (see above).
+
+- lcms is a new LCMS-based ICC color profile correction element
+
+- openmptdec is a new OpenMPT-based decoder for module music formats,
+ such as S3M, MOD, XM, IT. It is built on top of a new
+ GstNonstreamAudioDecoder base class which aims to unify handling of
+ files which do not operate a streaming model. The wildmidi plugin
+ has also been revived and is also implemented on top of this new
+ base class.
+
+- The curl plugin has gained a new curlhttpsrc element, which is
+ useful for testing HTTP protocol version 2.0 amongst other things.
+
+Noteworthy new API
+
+- GstPromise provides future/promise-like functionality. This is used
+ in the GStreamer WebRTC implementation.
+
+
+- GstReferenceTimestampMeta is a new meta that allows you to attach
+ additional reference timestamps to a buffer. These timestamps don't
+ have to relate to the pipeline clock in any way. Examples of this
+ could be an NTP timestamp when the media was captured, a frame
+ counter on the capture side or the (local) UNIX timestamp when the
+ media was captured. The decklink elements make use of this.
+
+
+- GstVideoRegionOfInterestMeta: it's now possible to attach generic
+ free-form element-specific parameters to a region of interest meta,
+ for example to tell a downstream encoder to use certain codec
+ parameters for a certain region.
+
+
+- gst_bus_get_pollfd can be used to obtain a file descriptor for the
+ bus that can be poll()-ed on for new messages. This is useful for
+ integration with non-GLib event loops.
+
+
+- gst_get_main_executable_path() can be used by wrapper plugins that
+ need to find things in the directory where the application
+ executable is located. In the same vein,
+ GST_PLUGIN_DEPENDENCY_FLAG_PATHS_ARE_RELATIVE_TO_EXE can be used to
+ signal that plugin dependency paths are relative to the main
+ executable.
+
+- pad templates can be told about the GType of the pad subclass of the
+ pad via newly-added GstPadTemplate API API or the
+ gst_element_class_add_static_pad_template_with_gtype() convenience
+ function. gst-inspect-1.0 will use this information to print pad
+ properties.
+
+
+- new convenience functions to iterate over element pads without using
+ the GstIterator API: gst_element_foreach_pad(),
+ gst_element_foreach_src_pad(), and gst_element_foreach_sink_pad().
+
+
+- GstBaseSrc and appsrc have gained support for buffer lists:
+ GstBaseSrc subclasses can use gst_base_src_submit_buffer_list(), and
+ applications can use gst_app_src_push_buffer_list() to push a buffer
+ list into appsrc.
+
+
+- The GstHarness unit test harness has a couple of new convenience
+ functions to retrieve all pending data in the harness in form of a
+ single chunk of memory.
+
+
+- GstAudioStreamAlign is a new helper object for audio elements that
+ handles discontinuity detection and sample alignment. It will align
+ samples after the previous buffer's samples, but keep track of the
+ divergence between buffer timestamps and sample position (jitter).
+ If it exceeds a configurable threshold the alignment will be reset.
+ This simply factors out code that was duplicated in a number of
+ elements into a common helper API.
+
+
+- The GstVideoEncoder base class implements Quality of Service (QoS)
+ now. This is disabled by default and must be opted in by setting the
+ "qos" property, which will make the base class gather statistics
+ about the real-time performance of the pipeline from downstream
+ elements (usually sinks that sync the pipeline clock). Subclasses
+ can then make use of this by checking whether input frames are late
+ already using gst_video_encoder_get_max_encode_time() If late, they
+ can just drop them and skip encoding in the hope that the pipeline
+ will catch up.
+
+
+- The GstVideoOverlay interface gained a few helper functions for
+ installing and handling a "render-rectangle" property on elements
+ that implement this interface, so that this functionality can also
+ be used from the command line for testing and debugging purposes.
+ The property wasn't added to the interface itself as that would
+ require all implementors to provide it which would not be
+ backwards-compatible.
+
+
+- A new base class, GstNonstreamAudioDecoder for non-stream audio
+ decoders was added to gst-plugins-bad. This base-class is meant to
+ be used for audio decoders that require the whole stream to be
+ loaded first before decoding can start. Examples of this are module
+ formats (MOD/S3M/XM/IT/etc), C64 SID tunes, video console music
+ files (GYM/VGM/etc), MIDI files and others. The new openmptdec
+ element is based on this.
+
+
+- Full list of API new in 1.14:
+- GStreamer core API new in 1.14
+- GStreamer base library API new in 1.14
+- gst-plugins-base libraries API new in 1.14
+- gst-plugins-bad: no list, mostly GstWebRTC library and new
+ non-stream audio decoder base class.
+
+New RTP features and improvements
+
+- rtpulpfecenc and rtpulpfecdec are new elements that implement
+ Generic Forward Error Correction (FEC) using Uneven Level Protection
+ (ULP) as described in RFC 5109. This can be used to protect against
+ certain types of (non-bursty) packet loss, and important packets
+ such as those containing codec configuration data or key frames can
+ be protected with higher redundancy. Equally, packets that are not
+ particularly important can be given low priority or not be protected
+ at all. If packets are lost, the receiver can then hopefully restore
+ the lost packet(s) from the surrounding packets which were received.
+ This is an alternative to, or rather complementary to, dealing with
+ packet loss using _retransmission (rtx)_. GStreamer has had
+ retransmission support for a long time, but Forward Error Correction
+ allows for different trade-offs: The advantage of Forward Error
+ Correction is that it doesn't add latency, whereas retransmission
+ requires at least one more roundtrip to request and hopefully
+ receive lost packets; Forward Error Correction increases the
+ required bandwidth however, even in situations where there is no
+ packet loss at all, so one will typically want to fine-tune the
+ overhead and mechanisms used based on the characteristics of the
+ link at the time.
+
+- New _Redundant Audio Data (RED)_ encoders and decoders for RTP as
+ per RFC 2198 are also provided (rtpredenc and rtpreddec), mostly for
+ chrome webrtc compatibility, as chrome will wrap ULPFEC-protected
+ streams in RED packets, and such streams need to be wrapped and
+ unwrapped in order to use ULPFEC with chrome.
+
+
+- a few new buffer flags for FEC support:
+ GST_BUFFER_FLAG_NON_DROPPABLE can be used to mark important buffers,
+ e.g. to flag RTP packets carrying keyframes or codec setup data for
+ RTP Forward Error Correction purposes, or to prevent still video
+ frames from being dropped by elements due to QoS. There already is a
+ GST_BUFFER_FLAG_DROPPABLE. GST_RTP_BUFFER_FLAG_REDUNDANT is used to
+ signal internally that a packet represents a redundant RTP packet
+ and used in rtpstorage to hold back the packet and use it only for
+ recovery from packet loss. Further work is still needed in
+ payloaders to make use of these.
+
+- rtpbin now has an option for increasing timestamp offsets gradually:
+ Instant large changes to the internal ts_offset may cause timestamps
+ to move backwards and also cause visible glitches in media playback.
+ The new "max-ts-offset-adjustment" and "max-ts-offset" properties
+ let the application control the rate to apply changes to ts_offset.
+ There have also been some EOS/BYE handling improvements in rtpbin.
+
+- rtpjitterbuffer has a new fast start mode: in many scenarios the
+ jitter buffer will have to wait for the full configured latency
+ before it can start outputting packets. The reason for that is that
+ it often can't know what the sequence number of the first expected
+ RTP packet is, so it can't know whether a packet earlier than the
+ earliest packet received will still arrive in future. This behaviour
+ can now be bypassed by setting the "faststart-min-packets" property
+ to the number of consecutive packets needed to start, and the jitter
+ buffer will start output packets as soon as it has N consecutive
+ packets queued internally. This is particularly useful to get a
+ first video frame decoded and rendered as quickly as possible.
+
+- rtpL8pay and rtpL8depay provide RTP payloading and depayloading for
+ 8-bit raw audio
+
+New element features
+
+- playbin3 has gained support or gapless playback via the
+ "about-to-finish" signal where users can set the uri for the next
+ item to play. For non-live streams this will be emitted as soon as
+ the first uri has finished downloading, so with sufficiently large
+ buffers it is now possible to pre-buffer the next item well ahead of
+ time (unlike playbin where there would not be a lot of time between
+ "about-to-finish" emission and the end of the stream). If the stream
+ format of the next stream is the same as that of the previous
+ stream, the data will be concatenated via the concat element.
+ Whether this will result in true gaplessness depends on the
+ container format and codecs used, there might still be codec-related
+ gaps between streams with some codecs.
+
+- tee now does allocation query aggregation, which is important for
+ zero-copy and efficient data handling, especially for video. Those
+ who want to drop allocation queries on purpose can use the identity
+ element's new "drop-allocation" property for that instead.
+
+- audioconvert now has a "mix-matrix" property, which obsoletes the
+ audiomixmatrix element. There's also mix matrix support in the audio
+ conversion and channel mixing API.
+
+- x264enc: new "insert-vui" property to disable VUI (Video Usability
+ Information) parameter insertion into the stream, which allows
+ creation of streams that are compatible with certain legacy hardware
+ decoders that will refuse to decode in certain combinations of
+ resolution and VUI parameters; the max. allowed number of B-frames
+ was also increased from 4 to 16.
+
+- dvdlpcmdec: has gained support for Blu-Ray audio LPCM.
+
+- appsrc has gained support for buffer lists (see above) and also seen
+ some other performance improvements.
+
+- flvmux has been ported to the GstAggregator base class which means
+ it can work in defined-latency mode with live input sources and
+ continue streaming if one of the inputs stops producing data.
+
+- jpegenc has gained a "snapshot" property just like pngenc to make it
+ easier to just output a single encoded frame.
+
+- jpegdec will now handle interlaced MJPEG streams properly and also
+ handle frames without an End of Image marker better.
+
+- v4l2: There are now video encoders for VP8, VP9, MPEG4, and H263.
+ The v4l2 video decoder handles dynamic resolution changes, and the
+ video4linux device provider now does much faster device probing. The
+ plugin also no longer uses the libv4l2 library by default, as it has
+ prevented a lot of interesting use cases like CREATE_BUFS, DMABuf,
+ usage of TRY_FMT. As the libv4l2 library is totally inactive and not
+ really maintained, we decided to disable it. This might affect a
+ small number of cheap/old webcams with custom vendor formats for
+ which we do not provide conversion in GStreamer. It is possible to
+ re-enable support for libv4l2 at run-time however, by setting the
+ environment variable GST_V4L2_USE_LIBV4L2=1.
+
+- rtspsrc now has support for RTSP protocol version 2.0 as well as
+ ONVIF audio backchannels (see below for more details). It also
+ sports a new ["accept-certificate"] signal for "manually" checking a
+ TLS certificate for validity. It now also prints RTSP/SDP messages
+ to the gstreamer debug log instead of stdout.
+
+- shout2send now uses non-blocking I/O and has a configurable network
+ operations timeout.
+
+- splitmuxsink has gained a "split-now" action signal and new
+ "alignment-threshold" and "use-robust-muxing" properties. If robust
+ muxing is enabled, it will check and set the muxer's reserved space
+ properties if present. This is primarily for use with mp4mux's
+ robust muxing mode.
+
+- qtmux has a new _prefill recording mode_ which sets up a moov header
+ with the correct sample positions beforehand, which then allows
+ software like Adobe Premiere and FinalCut Pro to import the files
+ while they are still being written to. This only works with constant
+ framerate I-frame only streams, and for now only support for ProRes
+ video and raw audio is implemented but adding new codecs is just a
+ matter of defining appropriate maximum frame sizes.
+
+- qtmux also supports writing of svmi atoms with stereoscopic video
+ information now. Trak timescales can be configured on a per-stream
+ basis using the "trak-timescale" property on the sink pads. Various
+ new formats can be muxed: MPEG layer 1 and 2, AC3 and Opus, as well
+ as PNG and VP9.
+
+- souphttpsrc now does connection sharing by default, shares its
+ SoupSession with other elements in the same pipeline via a
+ GstContext if possible (session-wide settings are all the defaults).
+ This allows for connection reuse, cookie sharing, etc. Applications
+ can also force a context to use. In other news, HTTP headers
+ received from the server are posted as element messages on the bus
+ now for easier diagnostics, and it's also possible now to use other
+ types of proxy servers such as SOCKS4 or SOCKS5 proxies, support for
+ which is implemented directly in gio. Before only HTTP proxies were
+ allowed.
+
+- qtmux, mp4mux and matroskamux will now refuse caps changes of input
+ streams at runtime. This isn't really supported with these
+ containers (or would have to be implemented differently with a
+ considerable effort) and doesn't produce valid and spec-compliant
+ files that will play everywhere. So if you can't guarantee that the
+ input caps won't change, use a container format that does support on
+ the fly caps changes for a stream such as MPEG-TS or use
+ splitmuxsink which can start a new file when the caps change. What
+ would happen before is that e.g. rtph264depay or rtph265depay would
+ simply send new SPS/PPS inband even for AVC format, which would then
+ get muxed into the container as if nothing changed. Some decoders
+ will handle this just fine, but that's often more luck than by
+ design. In any case, it's not right, so we disallow it now.
+
+- matroskamux had Table of Content (TOC) support now (chapters etc.)
+ and matroskademux TOC support has been improved. matroskademux has
+ also seen seeking improvements searching for the right cluster and
+ position.
+
+- videocrop now uses GstVideoCropMeta if downstream supports it, which
+ means cropping can be handled more efficiently without any copying.
+
+- compositor now has support for _crossfade blending_, which can be
+ used via the new "crossfade-ratio" property on the sink pads.
+
+- The avwait element has a new "end-timecode" property and posts
+ "avwait-status" element messages now whenever avwait starts or stops
+ passing through data (e.g. because target-timecode and end-timecode
+ respectively have been reached).
+
+
+- h265parse and h265parse will try harder to make upstream output the
+ same caps as downstream requires or prefers, thus avoiding
+ unnecessary conversion. The parsers also expose chroma format and
+ bit depth in the caps now.
+
+- The dtls elements now longer rely on or require the application to
+ run a GLib main loop that iterates the default main context
+ (GStreamer plugins should never rely on the application running a
+ GLib main loop).
+
+- openh264enc allows to change the encoding bitrate dynamically at
+ runtime now
+
+- nvdec is a new plugin for hardware-accelerated video decoding using
+ the NVIDIA NVDEC API (which replaces the old VDPAU API which is no
+ longer supported by NVIDIA)
+
+- The NVIDIA NVENC hardware-accelerated video encoders now support
+ dynamic bitrate and preset reconfiguration and support the I420
+ 4:2:0 video format. It's also possible to configure the gop size via
+ the new "gop-size" property.
+
+- The MPEG-TS muxer and demuxer (tsmux, tsdemux) now have support for
+ JPEG2000
+
+- openjpegdec and jpeg2000parse support 2-component images now (gray
+ with alpha), and jpeg2000parse has gained limited support for
+ conversion between JPEG2000 stream-formats. (JP2, J2C, JPC) and also
+ extracts more details such as colorimetry, interlace-mode,
+ field-order, multiview-mode and chroma siting.
+
+- The decklink plugin for Blackmagic capture and playback cards have
+ seen numerous improvements:
+
+- decklinkaudiosrc and decklinkvideosrc now put hardware reference
+ timestamp on buffers in form of GstReferenceTimestampMetas.
+ This can be useful to know on multi-channel cards which frames from
+ different channels were captured at the same time.
+
+- decklinkvideosink has gained support for Decklink hardware keying
+ with two new properties ("keyer-mode" and "keyer-level") to control
+ the built-in hardware keyer of Decklink cards.
+
+- decklinkaudiosink has been re-implemented around GstBaseSink instead
+ of the GstAudioBaseSink base class, since the Decklink APIs don't
+ fit very well with the GstAudioBaseSink APIs, which used to cause
+ various problems due to inaccuracies in the clock calculations.
+ Problems were audio drop-outs and A/V sync going wrong after
+ pausing/seeking.
+
+- support for more than 16 devices, without any artificial limit
+
+- work continued on the msdk plugin for Intel's Media SDK which
+ enables hardware-accelerated video encoding and decoding on Intel
+ graphics hardware on Windows or Linux. More tuning options were
+ added, and more pixel formats and video codecs are supported now.
+ The encoder now also handles force-key-unit events and can insert
+ frame-packing SEIs for side-by-side and top-bottom stereoscopic 3D
+ video.
+
+- dashdemux can now do adaptive trick play of certain types of DASH
+ streams, meaning it can do fast-forward/fast-rewind of normal (non-I
+ frame only) streams even at high speeds without saturating network
+ bandwidth or exceeding decoder capabilities. It will keep statistics
+ and skip keyframes or fragments as needed. See Sebastian's blog post
+ _DASH trick-mode playback in GStreamer_ for more details. It also
+ supports webvtt subtitle streams now and has seen improvements when
+ seeking in live streams.
+
+
+- kmssink has seen lots of fixes and improvements in this cycle,
+ including:
+
+- Raspberry Pi (vc4) and Xilinx DRM driver support
+
+- new "render-rectangle" property that can be used from the command
+ line as well as "display-width" and "display-height", and
+ "can-scale" properties
+
+- GstVideoCropMeta support
+
+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 fixed; some proprietary embedded Linux
+drivers don't work; macOS works).
+
+GstPhysMemoryAllocator interface moved from -bad to -base
+
+GstPhysMemoryAllocator is a marker interface for allocators with
+physical address backed memory.
+
+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.
-The video decoder is now able to negotiate the downstream format.
+- gst_util_dump_buffer() prints the content of a buffer to stdout.
-Elements that are dynamically created from /dev/video\* now track changes on
-these devices to ensure the registry stay up to date.
+- gst_pad_link_get_name() and gst_state_change_get_name() print pad
+ link return values and state change transition values as strings.
-All this and various bug fixes that improve both stability and correctness.
+- 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.
-### GStreamer Editing Services
+- Miniobject refcount tracing and logging was not entirley
+ thread-safe, there were duplicates or missing entries at times. This
+ has now been made reliable.
-Added APIs to handle asset proxying support. Proxy creation is not the
-responsibility of GES itself, but GES provides all the needed features
-for it to be cleanly handled at a higher level.
+- The netsim element, which can be used to simulate network jitter,
+ packet reordering and packet loss, received new features and
+ improvements: it can now also simulate network congestion using a
+ token bucket algorithm. This can be enabled via the "max-kbps"
+ property. Packet reordering can be disabled now via the
+ "allow-reordering" property: Reordering of packets is not very
+ common in networks, and the delay functions will always introduce
+ reordering if delay > packet-spacing, so by setting
+ "allow-reordering" to FALSE you guarantee that the packets are in
+ order, while at the same time introducing delay/jitter to them. By
+ using the new "delay-distribution" property the use can control how
+ the delay applied to delayed packets is distributed: This is either
+ the uniform distribution (as before) or the normal distribution; in
+ addition there is also the gamma distribution which simulates the
+ delay on wifi networks better.
-Added support for changing playback rate. This means that now, whenever a
-user adds a 'pitch' element (as it is the only known element to change playback
-rate through properties), GES will handle everything internally. This change
-introduced a new media-duration-factor property in NleObject which will
-lead to tweaking of seek events so they have the proper playback range to be
-requested upstream.
-Construction of NLE objects has been reworked making copy/pasting fully
-functional and allowing users to set properties on effects right after
-creating them.
+Tools
-Rework of the title source to add more flexibility in text positioning,
-and letting the user get feedback about rendered text positioning.
+- 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.
-Report nlecomposition structural issues (coming from user programing mistakes)
-into ERROR messages on the bus.
+- gst-launch-1.0 now generates a gstreamer pipeline diagram (.dot
+ file) whenever SIGHUP is sent to it on Linux/*nix systems.
-Add GI/pythyon testsuite in GES itself, making sure the API is working as expected
-in python, and allowing writing tests faster.
+- gst-discoverer-1.0 can now analyse live streams such as rtsp:// URIs
-### GstValidate
-Added support to run tests inside gdb.
+GStreamer RTSP server
-Added a 'smart' reporting mode where we give as much information as possible about
-critical errors.
+- Initial support for [RTSP protocol version
+ 2.0][rtsp2-lightning-talk] was added, which is to the best of our
+ knowledge the first RTSP 2.0 implementation ever!
-Uses GstTracer now instead of a LD\_PRELOAD library.
+- 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.
-## Miscellaneous
+
+- 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.
-- encodebin now works with "encoder-muxers" such as wavenc
+- rtspclientsink has gained an "accept-certificate" signal for
+ manually checking a TLS certificate for validity.
-- gst-play-1.0 acquired a new keyboard shortcut: '0' seeks back to the start
+- Fix keep-alive/timeout issue for certain clients using TCP
+ interleave as transport who don't do keep-alive via some other
+ method such as periodic RTSP OPTION requests. We now put netaddress
+ metas on the packets from the TCP interleaved stream, so can map
+ RTCP packets to the right stream in the server and can handle them
+ properly.
-- gst-play-1.0 supports two new command line switches: -v for verbose output
- and --flags to configure the playbin flags to use.
+- 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.
-## Build and Dependencies
+- Fix corner case issue where the wrong mount point was found when
+ there were multiple mount points with a common prefix.
-- The GLib dependency requirement was bumped to 2.40
-- The -Bsymbolic configure check now works with clang as well
+GStreamer VAAPI
-- ffmpeg is now required as libav provider, incompatible changes were
- introduced that make it no longer viable to support both FFmpeg and Libav
- as libav providers. Most major distros have switched to FFmpeg or are in
- the process of switching to it anyway, so we don't expect this to be a
- problem, and there is still an internal copy of ffmpeg that can be used
- as fallback if needed.
+- this section will be filled in shortly {FIXME!}
-- The internal ffmpeg snapshot is now FFMpeg 3.0, but it should be possible
- to build against 2.8 as well for the time being.
-## Platform-specific improvements
+GStreamer Editing Services and NLE
-### Android
+- this section will be filled in shortly {FIXME!}
-- Zero-copy video decoding on Android using the hardware-accelerated decoders
- has been implemented, and is fully integrated with the GStreamer OpenGL stack
-- ahcsrc, a new camera source element, has been merged and can be used to
- capture video on android devices. It uses the android.hardware.Camera Java
- API to capture from the system's cameras.
+GStreamer validate
-- The OpenGL-based QML video sink can now also be used on Android
+- this section will be filled in shortly {FIXME!}
-- New tinyalsasink element, which is mainly useful for Android but can also
- be used on other platforms.
-### OS/X and iOS
+GStreamer Python Bindings
-- The system clock now uses mach\_absolute\_time() on OSX/iOS, which is
- the preferred high-resolution monotonic clock to be used on Apple platforms
+- this section will be filled in shortly {FIXME!}
-- The OpenGL-based QML video sink can now also be used on OS X and iOS (with
- some Qt build system massaging)
-- New IOSurface based memory implementation in avfvideosrc and vtdec on OS X
- for zerocopy with OpenGL. The previously used OpenGL extension
- GL_APPLE_ycbcr_422 is not compatible with GL 3.x core contexts.
+Build and Dependencies
-- New GstAppleCoreVideoMemory wrapping CVPixelBuffer's
+- 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.
-- avfvideosrc now supports renegotiation.
+- gst-plugins-bad no longer depends on the libschroedinger Dirac codec
+ library.
-### Windows
+- The srtp plugin can now also be built against libsrtp2.
-- Various bugs with UDP and multicast were fixed on Windows, mostly related to
- gst-rtsp-server.
+- 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.
-- A few bugs in directsoundsrc and directsoundsink were fixed that could cause
- the element to lock up. Also the "mute" property on the sink was fixed, and
- a new "device" property for device selection was added to the source.
+- 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.
-## Known Issues
+- 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.
-- Building GStreamer applications with the Android NDK r11 is currently not
- supported due to incompatible changes in the NDK. This is expected to be
- fixed for 1.8.1.
- [Bugzilla #763999](https://bugzilla.gnome.org/show_bug.cgi?id=763999)
-- vp8enc crashes on 32 bit Windows, but was working fine in 1.6. 64 bit
- Windows is unaffected.
- [Bugzilla #763663](https://bugzilla.gnome.org/show_bug.cgi?id=763663)
+Note to packagers and distributors
-## Contributors
+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.
-Adam Miartus, Alban Bedel, Aleix Conchillo Flaqué, Aleksander Wabik,
-Alessandro Decina, Alex Ashley, Alex Dizengof, Alex Henrie, Alistair Buxton,
-Andreas Cadhalpun, Andreas Frisch, André Draszik, Anthony G. Basile,
-Antoine Jacoutot, Anton Bondarenko, Antonio Ospite, Arjen Veenhuizen,
-Arnaud Vrac, Arun Raghavan, Athanasios Oikonomou, Aurélien Zanelli, Ben Iofel,
-Bob Holcomb, Branko Subasic, Carlos Rafael Giani, Chris Bass, Csaba Toth,
-Daniel Kamil Kozar, Danilo Cesar Lemes de Paula, Dave Craig, David Fernandez,
-David Schleef, David Svensson Fors, David Waring, David Wu, Duncan Palmer,
-Edward Hervey, Egor Zaharov, Etienne Peron, Eunhae Choi, Evan Callaway,
-Evan Nemerson, Fabian Orccon, Florent Thiéry, Florin Apostol, Frédéric Wang,
-George Kiagiadakis, George Yunaev, Göran Jönsson, Graham Leggett,
-Guillaume Desmottes, Guillaume Marquebielle, Haihua Hu, Havard Graff,
-Heinrich Fink, Holger Kaelberer, HoonHee Lee, Hugues Fruchet, Hyunil Park,
-Hyunjun Ko, Ilya Konstantinov, James Stevenson, Jan Alexander Steffens (heftig),
-Jan Schmidt, Jason Litzinger, Jens Georg, Jimmy Ohn, Joan Pau Beltran,
-Joe Gorse, John Chang, John Slade, Jose Antonio Santos Cadenas, Josep Torra,
-Julian Bouzas, Julien Isorce, Julien Moutte, Justin Kim, Kazunori Kobayashi,
-Koop Mast, Lim Siew Hoon, Linus Svensson, Lubosz Sarnecki, Luis de Bethencourt,
-Lukasz Forynski, Manasa Athreya, Marcel Holtmann, Marcin Kolny, Marcus Prebble,
-Mark Nauwelaerts, Maroš Ondrášek, Martin Kelly, Matej Knopp, Mathias Hasselmann,
-Mathieu Duponchelle, Matt Crane, Matthew Marsh, Matthew Waters, Matthieu Bouron,
-Mersad Jelacic, Michael Olbrich, Miguel París Díaz, Mikhail Fludkov,
-Mischa Spiegelmock, Nicola Murino, Nicolas Dufresne, Nicolas Huet,
-Nirbheek Chauhan, Ognyan Tonchev, Olivier Crête, Pablo Anton, Pankaj Darak,
-Paolo Pettinato, Patricia Muscalu, Paul Arzelier, Pavel Bludov, Perry Hung,
-Peter Korsgaard, Peter Seiderer, Petr Viktorin, Philippe Normand,
-Philippe Renon, Philipp Zabel, Philip Van Hoof, Philip Withnall, Piotr Drąg,
-plamot, Polochon\_street, Prashant Gotarne, Rajat Verma, Ramiro Polla,
-Ravi Kiran K N, Reynaldo H. Verdejo Pinochet, Robert Swain, Romain Picard,
-Roman Nowicki, Ross Burton, Ryan Hendrickson, Santiago Carot-Nemesio,
-Scott D Phillips, Sebastian Dröge, Sebastian Rasmussen, Sergey Borovkov,
-Seungha Yang, Sjors Gielen, Song Bing, Sreerenj Balachandran, Srimanta Panda,
-Stavros Vagionitis, Stefan Sauer, Steven Hoving, Stian Selnes, Suhwang Kim,
-Thiago Santos, Thibault Saunier, Thijs Vermeir, Thomas Bluemel, Thomas Roos,
-Thomas Vander Stichele, Tim-Philipp Müller, Tim Sheridan, Ting-Wei Lan,
-Tom Deseyn, Vanessa Chipirrás Navalón, Víctor Manuel Jáquez Leal,
-Vincent Dehors, Vincent Penquerc'h, Vineeth T M, Vivia Nikolaidou,
-Wang Xin-yu (王昕宇), William Manley, Wim Taymans, Wonchul Lee, Xavi Artigas,
-Xavier Claessens, Youness Alaoui,
+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 1.8
-More than [~700 bugs][bugs-fixed-in-1.8] have been fixed during
-the development of 1.8.
+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 1.6 branch and fixed there as well, all fixes that ended up in the
-1.6 branch are also included in 1.8.
+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.
+
-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
-[bugs-fixed-in-1.8]: https://bugzilla.gnome.org/buglist.cgi?bug_status=RESOLVED&bug_status=VERIFIED&classification=Platform&limit=0&list_id=107311&order=bug_id&product=GStreamer&query_format=advanced&resolution=FIXED&target_milestone=1.6.1&target_milestone=1.6.2&target_milestone=1.6.3&target_milestone=1.7.0&target_milestone=1.7.1&target_milestone=1.7.2&target_milestone=1.7.3&target_milestone=1.7.4&target_milestone=1.7.90&target_milestone=1.7.91&target_milestone=1.7.92&target_milestone=1.7.x&target_milestone=1.8.0
+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.
-## Stable 1.8 branch
+1.14.0
-After the 1.8.0 release there will be several 1.8.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 1.8.x bug-fix releases will be made from the git 1.8 branch, which
-is a stable branch.
+1.14.0 is scheduled to be released in early March 2018.
-### 1.8.0
-1.8.0 was released on 24 March 2016.
+Known Issues
-### 1.8.1
+- 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.
-The first 1.8 bug-fix release (1.8.1) is planned for April 2016.
-## Schedule for 1.10
+Schedule for 1.16
-Our next major feature release will be 1.10, and 1.9 will be the unstable
-development version leading up to the stable 1.10 release. The development
-of 1.9/1.10 will happen in the git master branch.
+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 1.10 development cycle is yet to be confirmed, but it is
-expected that feature freeze will be around late July or early August,
-followed by several 1.9 pre-releases and the new 1.10 stable release
-in September.
+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.
-1.10 will be backwards-compatible to the stable 1.8, 1.6, 1.4, 1.2 and 1.0
-release series.
+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, Nicolas Dufresne, Edward Hervey, Víctor
-Manuel Jáquez Leal, Arun Raghavan, Thiago Santos, Thibault Saunier, Jan
-Schmidt and Matthew Waters.*
+_These release notes have been prepared by Tim-Philipp Müller with_
+_contributions from Sebastian Dröge._
-*License: [CC BY-SA 4.0](http://creativecommons.org/licenses/by-sa/4.0/)*
+_License: CC BY-SA 4.0_
projects / platform / upstream / gstreamer.git / blobdiff