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diff --git a/NEWS b/NEWS
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--- a/NEWS
+++ b/NEWS
@@ -1,21 +1,25 @@
 
 
-GSTREAMER 1.14 RELEASE NOTES
+GSTREAMER 1.16 RELEASE NOTES
 
 
-The GStreamer team is proud to announce a new major feature release in
-the stable 1.x API series of your favourite cross-platform multimedia
-framework!
+GStreamer 1.16 has not been released yet. It is scheduled for release
+around September 2018.
 
-As always, this release is again packed with new features, bug fixes and
-other improvements.
+1.15.0.1 is the unstable development version that is being developed in
+the git master branch and which will eventually result in 1.16.
+
+The plan for the 1.16 development cycle is yet to be confirmed, but it
+is expected that feature freeze will be around August 2017 followed by
+several 1.15 pre-releases and the new 1.16 stable release in September.
 
-GStreamer 1.14.0 was released on 19 March 2018.
+1.16 will be backwards-compatible to the stable 1.14, 1.12, 1.10, 1.8,
+1.6, 1.4, 1.2 and 1.0 release series.
 
-See https://gstreamer.freedesktop.org/releases/1.14/ for the latest
+See https://gstreamer.freedesktop.org/releases/1.16/ for the latest
 version of this document.
 
-_Last updated: Monday 19 March 2018, 12:00 UTC (log)_
+_Last updated: Tuesday 20 March 2018, 01:30 UTC (log)_
 
 
 Introduction
@@ -30,1165 +34,154 @@ other improvements.
 
 Highlights
 
--   WebRTC support: real-time audio/video streaming to and from web
-    browsers
-
--   Experimental support for the next-gen royalty-free AV1 video codec
-
--   Video4Linux: encoding support, stable element names and faster
-    device probing
-
--   Support for the Secure Reliable Transport (SRT) video streaming
-    protocol
-
--   RTP Forward Error Correction (FEC) support (ULPFEC)
-
--   RTSP 2.0 support in rtspsrc and gst-rtsp-server
-
--   ONVIF audio backchannel support in gst-rtsp-server and rtspsrc
-
--   playbin3 gapless playback and pre-buffering support
-
--   tee, our stream splitter/duplication element, now does allocation
-    query aggregation which is important for efficient data handling and
-    zero-copy
-
--   QuickTime muxer has a new prefill recording mode that allows file
-    import in Adobe Premiere and FinalCut Pro while the file is still
-    being written.
-
--   rtpjitterbuffer fast-start mode and timestamp offset adjustment
-    smoothing
-
--   souphttpsrc connection sharing, which allows for connection reuse,
-    cookie sharing, etc.
-
--   nvdec: new plugin for hardware-accelerated video decoding using the
-    NVIDIA NVDEC API
-
--   Adaptive DASH trick play support
-
--   ipcpipeline: new plugin that allows splitting a pipeline across
-    multiple processes
-
--   Major gobject-introspection annotation improvements for large parts
-    of the library API
-
--   GStreamer C# bindings have been revived and seen many updates and
-    fixes
-
--   The externally maintained GStreamer Rust bindings had many usability
-    improvements and cover most of the API now. Coinciding with the 1.14
-    release, a new release with the 1.14 API additions is happening.
+-   this section will be completed in due course
 
 
 Major new features and changes
 
-WebRTC support
-
-There is now basic support for WebRTC in GStreamer in form of a new
-webrtcbin element and a webrtc support library. This allows you to build
-applications that set up connections with and stream to and from other
-WebRTC peers, whilst leveraging all of the usual GStreamer features such
-as hardware-accelerated encoding and decoding, OpenGL integration,
-zero-copy and embedded platform support. And it's easy to build and
-integrate into your application too!
-
-WebRTC enables real-time communication of audio, video and data with web
-browsers and native apps, and it is supported or about to be support by
-recent versions of all major browsers and operating systems.
-
-GStreamer's new WebRTC implementation uses libnice for Interactive
-Connectivity Establishment (ICE) to figure out the best way to
-communicate with other peers, punch holes into firewalls, and traverse
-NATs.
-
-The implementation is not complete, but all the basics are there, and
-the code sticks fairly close to the PeerConnection API. Where
-functionality is missing it should be fairly obvious where it needs to
-go.
+Noteworthy new API
 
-For more details, background and example code, check out Nirbheek's blog
-post _GStreamer has grown a WebRTC implementation_, as well as Matthew's
-_GStreamer WebRTC_ talk from last year's GStreamer Conference in Prague.
+-   this section will be filled in in due course
 
 New Elements
 
--   webrtcbin handles the transport aspects of webrtc connections (see
-    WebRTC section above for more details)
-
--   New srtsink and srtsrc elements for the Secure Reliable Transport
-    (SRT) video streaming protocol, which aims to be easy to use whilst
-    striking a new balance between reliability and latency for low
-    latency video streaming use cases. More details about SRT and the
-    implementation in GStreamer in Olivier's blog post _SRT in
-    GStreamer_.
-
--   av1enc and av1dec elements providing experimental support for the
-    next-generation royalty free video AV1 codec, alongside Matroska
-    support for it.
-
--   hlssink2 is a rewrite of the existing hlssink element, but unlike
-    its predecessor hlssink2 takes elementary streams as input and
-    handles the muxing to MPEG-TS internally. It also leverages
-    splitmuxsink internally to do the splitting. This allows more
-    control over the chunk splitting and sizing process and relies less
-    on the co-operation of an upstream muxer. Different to the old
-    hlssink it also works with pre-encoded streams and does not require
-    close interaction with an upstream encoder element.
-
--   audiolatency is a new element for measuring audio latency end-to-end
-    and is useful to measure roundtrip latency including both the
-    GStreamer-internal latency as well as latency added by external
-    components or circuits.
-
--   'fakevideosink is basically a null sink for video data and very
-    similar to fakesink, only that it will answer allocation queries and
-    will advertise support for various video-specific things such
-    GstVideoMeta, GstVideoCropMeta and GstVideoOverlayCompositionMeta
-    like a normal video sink would. This is useful for throughput
-    testing and testing the zero-copy path when creating a new pipeline.
-
--   ipcpipeline: new plugin that allows the splitting of a pipeline into
-    multiple processes. Usually a GStreamer pipeline runs in a single
-    process and parallelism is achieved by distributing workloads using
-    multiple threads. This means that all elements in the pipeline have
-    access to all the other elements' memory space however, including
-    that of any libraries used. For security reasons one might therefore
-    want to put sensitive parts of a pipeline such as DRM and decryption
-    handling into a separate process to isolate it from the rest of the
-    pipeline. This can now be achieved with the new ipcpipeline plugin.
-    Check out George's blog post _ipcpipeline: Splitting a GStreamer
-    pipeline into multiple processes_ or his lightning talk from last
-    year's GStreamer Conference in Prague for all the gory details.
-
--   proxysink and proxysrc are new elements to pass data from one
-    pipeline to another within the same process, very similar to the
-    existing inter elements, but not limited to raw audio and video
-    data. These new proxy elements are very special in how they work
-    under the hood, which makes them extremely powerful, but also
-    dangerous if not used with care. The reason for this is that it's
-    not just data that's passed from sink to src, but these elements
-    basically establish a two-way wormhole that passes through queries
-    and events in both directions, which means caps negotiation and
-    allocation query driven zero-copy can work through this wormhole.
-    There are scheduling considerations as well: proxysink forwards
-    everything into the proxysrc pipeline directly from the proxysink
-    streaming thread. There is a queue element inside proxysrc to
-    decouple the source thread from the sink thread, but that queue is
-    not unlimited, so it is entirely possible that the proxysink
-    pipeline thread gets stuck in the proxysrc pipeline, e.g. when that
-    pipeline is paused or stops consuming data for some other reason.
-    This means that one should always shut down down the proxysrc
-    pipeline before shutting down the proxysink pipeline, for example.
-    Or at least take care when shutting down pipelines. Usually this is
-    not a problem though, especially not in live pipelines. For more
-    information see Nirbheek's blog post _Decoupling GStreamer
-    Pipelines_, and also check out out the new ipcpipeline plugin for
-    sending data from one process to another process (see above).
-
--   lcms is a new LCMS-based ICC color profile correction element
-
--   openmptdec is a new OpenMPT-based decoder for module music formats,
-    such as S3M, MOD, XM, IT. It is built on top of a new
-    GstNonstreamAudioDecoder base class which aims to unify handling of
-    files which do not operate a streaming model. The wildmidi plugin
-    has also been revived and is also implemented on top of this new
-    base class.
-
--   The curl plugin has gained a new curlhttpsrc element, which is
-    useful for testing HTTP protocol version 2.0 amongst other things.
-
--   The msdk plugin has gained a MPEG-2 video decoder(msdkmpeg2dec), VP8
-    decoder(msdkvp8dec) and a VC1/WMV decoder(msdkvc1dec)
+-   this section will be filled in in due course
 
-Noteworthy new API
+New element features and additions
 
--   GstPromise provides future/promise-like functionality. This is used
-    in the GStreamer WebRTC implementation.
-
--   GstReferenceTimestampMeta is a new meta that allows you to attach
-    additional reference timestamps to a buffer. These timestamps don't
-    have to relate to the pipeline clock in any way. Examples of this
-    could be an NTP timestamp when the media was captured, a frame
-    counter on the capture side or the (local) UNIX timestamp when the
-    media was captured. The decklink elements make use of this.
-
--   GstVideoRegionOfInterestMeta: it's now possible to attach generic
-    free-form element-specific parameters to a region of interest meta,
-    for example to tell a downstream encoder to use certain codec
-    parameters for a certain region.
-
--   gst_bus_get_pollfd can be used to obtain a file descriptor for the
-    bus that can be poll()-ed on for new messages. This is useful for
-    integration with non-GLib event loops.
-
--   gst_get_main_executable_path() can be used by wrapper plugins that
-    need to find things in the directory where the application
-    executable is located. In the same vein,
-    GST_PLUGIN_DEPENDENCY_FLAG_PATHS_ARE_RELATIVE_TO_EXE can be used to
-    signal that plugin dependency paths are relative to the main
-    executable.
-
--   pad templates can be told about the GType of the pad subclass of the
-    pad via newly-added GstPadTemplate API API or the
-    gst_element_class_add_static_pad_template_with_gtype() convenience
-    function. gst-inspect-1.0 will use this information to print pad
-    properties.
-
--   new convenience functions to iterate over element pads without using
-    the GstIterator API: gst_element_foreach_pad(),
-    gst_element_foreach_src_pad(), and gst_element_foreach_sink_pad().
-
--   GstBaseSrc and appsrc have gained support for buffer lists:
-    GstBaseSrc subclasses can use gst_base_src_submit_buffer_list(), and
-    applications can use gst_app_src_push_buffer_list() to push a buffer
-    list into appsrc.
-
--   The GstHarness unit test harness has a couple of new convenience
-    functions to retrieve all pending data in the harness in form of a
-    single chunk of memory.
-
--   GstAudioStreamAlign is a new helper object for audio elements that
-    handles discontinuity detection and sample alignment. It will align
-    samples after the previous buffer's samples, but keep track of the
-    divergence between buffer timestamps and sample position (jitter).
-    If it exceeds a configurable threshold the alignment will be reset.
-    This simply factors out code that was duplicated in a number of
-    elements into a common helper API.
-
--   The GstVideoEncoder base class implements Quality of Service (QoS)
-    now. This is disabled by default and must be opted in by setting the
-    "qos" property, which will make the base class gather statistics
-    about the real-time performance of the pipeline from downstream
-    elements (usually sinks that sync the pipeline clock). Subclasses
-    can then make use of this by checking whether input frames are late
-    already using gst_video_encoder_get_max_encode_time() If late, they
-    can just drop them and skip encoding in the hope that the pipeline
-    will catch up.
-
--   The GstVideoOverlay interface gained a few helper functions for
-    installing and handling a "render-rectangle" property on elements
-    that implement this interface, so that this functionality can also
-    be used from the command line for testing and debugging purposes.
-    The property wasn't added to the interface itself as that would
-    require all implementors to provide it which would not be
-    backwards-compatible.
-
--   A new base class, GstNonstreamAudioDecoder for non-stream audio
-    decoders was added to gst-plugins-bad. This base-class is meant to
-    be used for audio decoders that require the whole stream to be
-    loaded first before decoding can start. Examples of this are module
-    formats (MOD/S3M/XM/IT/etc), C64 SID tunes, video console music
-    files (GYM/VGM/etc), MIDI files and others. The new openmptdec
-    element is based on this.
-
--   Full list of API new in 1.14:
--   GStreamer core API new in 1.14
--   GStreamer base library API new in 1.14
--   gst-plugins-base libraries API new in 1.14
--   gst-plugins-bad: no list, mostly GstWebRTC library and new
-    non-stream audio decoder base class.
-
-New RTP features and improvements
-
--   rtpulpfecenc and rtpulpfecdec are new elements that implement
-    Generic Forward Error Correction (FEC) using Uneven Level Protection
-    (ULP) as described in RFC 5109. This can be used to protect against
-    certain types of (non-bursty) packet loss, and important packets
-    such as those containing codec configuration data or key frames can
-    be protected with higher redundancy. Equally, packets that are not
-    particularly important can be given low priority or not be protected
-    at all. If packets are lost, the receiver can then hopefully restore
-    the lost packet(s) from the surrounding packets which were received.
-    This is an alternative to, or rather complementary to, dealing with
-    packet loss using _retransmission (rtx)_. GStreamer has had
-    retransmission support for a long time, but Forward Error Correction
-    allows for different trade-offs: The advantage of Forward Error
-    Correction is that it doesn't add latency, whereas retransmission
-    requires at least one more roundtrip to request and hopefully
-    receive lost packets; Forward Error Correction increases the
-    required bandwidth however, even in situations where there is no
-    packet loss at all, so one will typically want to fine-tune the
-    overhead and mechanisms used based on the characteristics of the
-    link at the time.
-
--   New _Redundant Audio Data (RED)_ encoders and decoders for RTP as
-    per RFC 2198 are also provided (rtpredenc and rtpreddec), mostly for
-    chrome webrtc compatibility, as chrome will wrap ULPFEC-protected
-    streams in RED packets, and such streams need to be wrapped and
-    unwrapped in order to use ULPFEC with chrome.
-
--   a few new buffer flags for FEC support:
-    GST_BUFFER_FLAG_NON_DROPPABLE can be used to mark important buffers,
-    e.g. to flag RTP packets carrying keyframes or codec setup data for
-    RTP Forward Error Correction purposes, or to prevent still video
-    frames from being dropped by elements due to QoS. There already is a
-    GST_BUFFER_FLAG_DROPPABLE. GST_RTP_BUFFER_FLAG_REDUNDANT is used to
-    signal internally that a packet represents a redundant RTP packet
-    and used in rtpstorage to hold back the packet and use it only for
-    recovery from packet loss. Further work is still needed in
-    payloaders to make use of these.
-
--   rtpbin now has an option for increasing timestamp offsets gradually:
-    Sudden large changes to the internal ts_offset may cause timestamps
-    to move backwards and may also cause visible glitches in media
-    playback. The new "max-ts-offset-adjustment" and "max-ts-offset"
-    properties let the application control the rate to apply changes to
-    ts_offset. There have also been some EOS/BYE handling improvements
-    in rtpbin.
-
--   rtpjitterbuffer has a new fast start mode: in many scenarios the
-    jitter buffer will have to wait for the full configured latency
-    before it can start outputting packets. The reason for that is that
-    it often can't know what the sequence number of the first expected
-    RTP packet is, so it can't know whether a packet earlier than the
-    earliest packet received will still arrive in future. This behaviour
-    can now be bypassed by setting the "faststart-min-packets" property
-    to the number of consecutive packets needed to start, and the jitter
-    buffer will start output packets as soon as it has N consecutive
-    packets queued internally. This is particularly useful to get a
-    first video frame decoded and rendered as quickly as possible.
-
--   rtpL8pay and rtpL8depay provide RTP payloading and depayloading for
-    8-bit raw audio
-
-New element features
-
--   playbin3 has gained support or gapless playback via the
-    "about-to-finish" signal where users can set the uri for the next
-    item to play. For non-live streams this will be emitted as soon as
-    the first uri has finished downloading, so with sufficiently large
-    buffers it is now possible to pre-buffer the next item well ahead of
-    time (unlike playbin where there would not be a lot of time between
-    "about-to-finish" emission and the end of the stream). If the stream
-    format of the next stream is the same as that of the previous
-    stream, the data will be concatenated via the concat element.
-    Whether this will result in true gaplessness depends on the
-    container format and codecs used, there might still be codec-related
-    gaps between streams with some codecs.
-
--   tee now does allocation query aggregation, which is important for
-    zero-copy and efficient data handling, especially for video. Those
-    who want to drop allocation queries on purpose can use the identity
-    element's new "drop-allocation" property for that instead.
-
--   audioconvert now has a "mix-matrix" property, which obsoletes the
-    audiomixmatrix element. There's also mix matrix support in the audio
-    conversion and channel mixing API.
-
--   x264enc: new "insert-vui" property to disable VUI (Video Usability
-    Information) parameter insertion into the stream, which allows
-    creation of streams that are compatible with certain legacy hardware
-    decoders that will refuse to decode in certain combinations of
-    resolution and VUI parameters; the max. allowed number of B-frames
-    was also increased from 4 to 16.
-
--   dvdlpcmdec: has gained support for Blu-Ray audio LPCM.
-
--   appsrc has gained support for buffer lists (see above) and also seen
-    some other performance improvements.
-
--   flvmux has been ported to the GstAggregator base class which means
-    it can work in defined-latency mode with live input sources and
-    continue streaming if one of the inputs stops producing data.
-
--   jpegenc has gained a "snapshot" property just like pngenc to make it
-    easier to output just a single encoded frame.
-
--   jpegdec will now handle interlaced MJPEG streams properly and also
-    handles frames without an End of Image marker better.
-
--   v4l2: There are now video encoders for VP8, VP9, MPEG4, and H263.
-    The v4l2 video decoder handles dynamic resolution changes, and the
-    video4linux device provider now does much faster device probing. The
-    plugin also no longer uses the libv4l2 library by default, as it has
-    prevented a lot of interesting use cases like CREATE_BUFS, DMABuf,
-    usage of TRY_FMT. As the libv4l2 library is totally inactive and not
-    really maintained, we decided to disable it. This might affect a
-    small number of cheap/old webcams with custom vendor formats for
-    which we do not provide conversion in GStreamer. It is possible to
-    re-enable support for libv4l2 at run-time however, by setting the
-    environment variable GST_V4L2_USE_LIBV4L2=1.
-
--   rtspsrc now has support for RTSP protocol version 2.0 as well as
-    ONVIF audio backchannels (see below for more details). It also
-    sports a new "accept-certificate" signal for "manually" checking a
-    TLS certificate for validity. It now also prints RTSP/SDP messages
-    to the gstreamer debug log instead of stdout.
-
--   shout2send now uses non-blocking I/O and has a configurable network
-    operations timeout.
-
--   splitmuxsink has gained a "split-now" action signal and new
-    "alignment-threshold" and "use-robust-muxing" properties. If robust
-    muxing is enabled, it will check and set the muxer's reserved space
-    properties if present. This is primarily for use with mp4mux's
-    robust muxing mode.
-
--   qtmux has a new _prefill recording mode_ which sets up a moov header
-    with the correct sample positions beforehand, which then allows
-    software like Adobe Premiere and FinalCut Pro to import the files
-    while they are still being written to. This only works with constant
-    framerate I-frame only streams, and for now only support for ProRes
-    video and raw audio is implemented. Adding support for additional
-    codecs is just a matter of defining appropriate maximum frame sizes
-    though.
-
--   qtmux also supports writing of svmi atoms with stereoscopic video
-    information now. Trak timescales can be configured on a per-stream
-    basis using the "trak-timescale" property on the sink pads. Various
-    new formats can be muxed: MPEG layer 1 and 2, AC3 and Opus, as well
-    as PNG and VP9.
-
--   souphttpsrc now does connection sharing by default: it shares its
-    SoupSession with other elements in the same pipeline via a
-    GstContext if possible (session-wide settings are all the defaults).
-    This allows for connection reuse, cookie sharing, etc. Applications
-    can also force a context to use. In other news, HTTP headers
-    received from the server are posted as element messages on the bus
-    now for easier diagnostics, and it's also possible now to use other
-    types of proxy servers such as SOCKS4 or SOCKS5 proxies, support for
-    which is implemented directly in gio. Before only HTTP proxies were
-    allowed.
-
--   qtmux, mp4mux and matroskamux will now refuse caps changes of input
-    streams at runtime. This isn't really supported with these
-    containers (or would have to be implemented differently with a
-    considerable effort) and doesn't produce valid and spec-compliant
-    files that will play everywhere. So if you can't guarantee that the
-    input caps won't change, use a container format that does support on
-    the fly caps changes for a stream such as MPEG-TS or use
-    splitmuxsink which can start a new file when the caps change. What
-    would happen before is that e.g. rtph264depay or rtph265depay would
-    simply send new SPS/PPS inband even for AVC format, which would then
-    get muxed into the container as if nothing changed. Some decoders
-    will handle this just fine, but that's often more luck than by
-    design. In any case, it's not right, so we disallow it now.
-
--   matroskamux has Table of Content (TOC) support now (chapters etc.)
-    and matroskademux TOC support has been improved. matroskademux has
-    also seen seeking improvements searching for the right cluster and
-    position.
-
--   videocrop now uses GstVideoCropMeta if downstream supports it, which
-    means cropping can be handled more efficiently without any copying.
-
--   compositor now has support for _crossfade blending_, which can be
-    used via the new "crossfade-ratio" property on the sink pads.
-
--   The avwait element has a new "end-timecode" property and posts
-    "avwait-status" element messages now whenever avwait starts or stops
-    passing through data (e.g. because target-timecode and end-timecode
-    respectively have been reached).
-
--   h265parse and h265parse will try harder to make upstream output the
-    same caps as downstream requires or prefers, thus avoiding
-    unnecessary conversion. The parsers also expose chroma format and
-    bit depth in the caps now.
-
--   The dtls elements now longer rely on or require the application to
-    run a GLib main loop that iterates the default main context
-    (GStreamer plugins should never rely on the application running a
-    GLib main loop).
-
--   openh264enc allows to change the encoding bitrate dynamically at
-    runtime now
-
--   nvdec is a new plugin for hardware-accelerated video decoding using
-    the NVIDIA NVDEC API (which replaces the old VDPAU API which is no
-    longer supported by NVIDIA)
-
--   The NVIDIA NVENC hardware-accelerated video encoders now support
-    dynamic bitrate and preset reconfiguration and support the I420
-    4:2:0 video format. It's also possible to configure the gop size via
-    the new "gop-size" property.
-
--   The MPEG-TS muxer and demuxer (tsmux, tsdemux) now have support for
-    JPEG2000
-
--   openjpegdec and jpeg2000parse support 2-component images now (gray
-    with alpha), and jpeg2000parse has gained limited support for
-    conversion between JPEG2000 stream-formats. (JP2, J2C, JPC) and also
-    extracts more details such as colorimetry, interlace-mode,
-    field-order, multiview-mode and chroma siting.
-
--   The decklink plugin for Blackmagic capture and playback cards have
-    seen numerous improvements:
-
--   decklinkaudiosrc and decklinkvideosrc now put hardware reference
-    timestamp on buffers in form of GstReferenceTimestampMetas.
-    This can be useful to know on multi-channel cards which frames from
-    different channels were captured at the same time.
-
--   decklinkvideosink has gained support for Decklink hardware keying
-    with two new properties ("keyer-mode" and "keyer-level") to control
-    the built-in hardware keyer of Decklink cards.
-
--   decklinkaudiosink has been re-implemented around GstBaseSink instead
-    of the GstAudioBaseSink base class, since the Decklink APIs don't
-    fit very well with the GstAudioBaseSink APIs, which used to cause
-    various problems due to inaccuracies in the clock calculations.
-    Problems were audio drop-outs and A/V sync going wrong after
-    pausing/seeking.
-
--   support for more than 16 devices, without any artificial limit
-
--   work continued on the msdk plugin for Intel's Media SDK which
-    enables hardware-accelerated video encoding and decoding on Intel
-    graphics hardware on Windows or Linux. Added the video memory,
-    buffer pool, and context/session sharing support which helps to
-    improve the performance and resource utilization. Rendernode support
-    is in place which helps to avoid the constraint of having a running
-    graphics server as DRM-Master. Encoders are exposing a number rate
-    control algorithms now. More encoder tuning options like
-    trellis-quantiztion (h264), slice size control (h264), B-pyramid
-    prediction(h264), MB-level bitrate control, frame partitioning and
-    adaptive I/B frame insertion were added, and more pixel formats and
-    video codecs are supported now. The encoder now also handles
-    force-key-unit events and can insert frame-packing SEIs for
-    side-by-side and top-bottom stereoscopic 3D video.
-
--   dashdemux can now do adaptive trick play of certain types of DASH
-    streams, meaning it can do fast-forward/fast-rewind of normal (non-I
-    frame only) streams even at high speeds without saturating network
-    bandwidth or exceeding decoder capabilities. It will keep statistics
-    and skip keyframes or fragments as needed. See Sebastian's blog post
-    _DASH trick-mode playback in GStreamer_ for more details. It also
-    supports webvtt subtitle streams now and has seen improvements when
-    seeking in live streams.
-
--   kmssink has seen lots of fixes and improvements in this cycle,
-    including:
-
--   Raspberry Pi (vc4) and Xilinx DRM driver support
-
--   new "render-rectangle" property that can be used from the command
-    line as well as "display-width" and "display-height", and
-    "can-scale" properties
-
--   GstVideoCropMeta support
+-   this section will be filled in in due course
 
 Plugin and library moves
 
-MPEG-1 audio (mp1, mp2, mp3) decoders and encoders moved to -good
-
-Following the expiration of the last remaining mp3 patents in most
-jurisdictions, and the termination of the mp3 licensing program, as well
-as the decision by certain distros to officially start shipping full mp3
-decoding and encoding support, these plugins should now no longer be
-problematic for most distributors and have therefore been moved from
--ugly and -bad to gst-plugins-good. Distributors can still disable these
-plugins if desired.
-
-In particular these are:
-
--   mpg123audiodec: an mp1/mp2/mp3 audio decoder using libmpg123
--   lamemp3enc: an mp3 encoder using LAME
--   twolamemp2enc: an mp2 encoder using TwoLAME
-
-GstAggregator moved from -bad to core
-
-GstAggregator has been moved from gst-plugins-bad to the base library in
-GStreamer and is now stable API.
-
-GstAggregator is a new base class for mixers and muxers that have to
-handle multiple input pads and aggregate streams into one output stream.
-It improves upon the existing GstCollectPads API in that it is a proper
-base class which was also designed with live streaming in mind.
-GstAggregator subclasses will operate in a mode with defined latency if
-any of the inputs are live streams. This ensures that the pipeline won't
-stall if any of the inputs stop producing data, and that the configured
-maximum latency is never exceeded.
-
-GstAudioAggregator, audiomixer and audiointerleave moved from -bad to -base
-
-GstAudioAggregator is a new base class for raw audio mixers and muxers
-and is based on GstAggregator (see above). It provides defined-latency
-mixing of raw audio inputs and ensures that the pipeline won't stall
-even if one of the input streams stops producing data.
-
-As part of the move to stabilise the API there were some last-minute API
-changes and clean-ups, but those should mostly affect internal elements.
-
-It is used by the audiomixer element, which is a replacement for
-'adder', which did not handle live inputs very well and did not align
-input streams according to running time. audiomixer should behave much
-better in that respect and generally behave as one would expected in
-most scenarios.
-
-Similarly, audiointerleave replaces the 'interleave' element which did
-not handle live inputs or non-aligned inputs very robustly.
-
-GstAudioAggregator and its subclases have gained support for input
-format conversion, which does not include sample rate conversion though
-as that would add additional latency. Furthermore, GAP events are now
-handled correctly.
-
-We hope to move the video equivalents (GstVideoAggregator and
-compositor) to -base in the next cycle, i.e. for 1.16.
-
-GStreamer OpenGL integration library and plugin moved from -bad to -base
-
-The GStreamer OpenGL integration library and opengl plugin have moved
-from gst-plugins-bad to -base and are now part of the stable API canon.
-Not all OpenGL elements have been moved; a few had to be left behind in
-gst-plugins-bad in the new openglmixers plugin, because they depend on
-the GstVideoAggregator base class which we were not able to move in this
-cycle. We hope to reunite these elements with the rest of their family
-for 1.16 though.
-
-This is quite a milestone, thanks to everyone who worked to make this
-happen!
-
-Qt QML and GTK plugins moved from -bad to -good
-
-The Qt QML-based qmlgl plugin has moved to -good and provides a
-qmlglsink video sink element as well as a qmlglsrc element. qmlglsink
-renders video into a QQuickItem, and qmlglsrc captures a window from a
-QML view and feeds it as video into a pipeline for further processing.
-Both elements leverage GStreamer's OpenGL integration. In addition to
-the move to -good the following features were added:
-
--   A proxy object is now used for thread-safe access to the QML widget
-    which prevents crashes in corner case scenarios: QML can destroy the
-    video widget at any time, so without this we might be left with a
-    dangling pointer.
-
--   EGL is now supported with the X11 backend, which works e.g. on
-    Freescale imx6
-
-The GTK+ plugin has also moved from -bad to -good. It includes gtksink
-and gtkglsink which both render video into a GtkWidget. gtksink uses
-Cairo for rendering the video, which will work everywhere in all
-scenarios but involves an extra memory copy, whereas gtkglsink fully
-leverages GStreamer's OpenGL integration, but might not work properly in
-all scenarios, e.g. where the OpenGL driver does not properly support
-multiple sharing contexts in different threads; on Linux Nouveau is
-known to be broken in this respect, whilst NVIDIA's proprietary drivers
-and most other drivers generally work fine, and the experience with
-Intel's driver seems to be mixed; some proprietary embedded Linux
-drivers don't work; macOS works).
-
-GstPhysMemoryAllocator interface moved from -bad to -base
-
-GstPhysMemoryAllocator is a marker interface for allocators with
-physical address backed memory.
+-   this section will be filled in in due course
 
 Plugin removals
 
--   the sunaudio plugin was removed, since it couldn't ever have been
-    built or used with GStreamer 1.0, but no one even noticed in all
-    these years.
+-   this section will be filled in in due course
 
--   the schroedinger-based Dirac encoder/decoder plugin has been
-    removed, as there is no longer any upstream or anyone else
-    maintaining it. Seeing that it's quite a fringe codec it seemed best
-    to simply remove it.
 
-API removals
+Miscellaneous API additions
 
--   some MPEG video parser API in the API unstable codecutils library in
-    gst-plugins-bad was removed after having been deprecated for 5
-    years.
+-   this section will be filled in in due course
 
+GstPlayer
 
-Miscellaneous changes
+-   this section will be filled in in due course
 
--   The video support library has gained support for a few new pixel
-    formats:
--   NV16_10LE32: 10-bit variant of NV16, packed into 32bit words (plus 2
-    bits padding)
--   NV12_10LE32: 10-bit variant of NV12, packed into 32bit words (plus 2
-    bits padding)
--   GRAY10_LE32: 10-bit grayscale, packed in 32bit words (plus 2 bits
-    padding)
-
--   decodebin, playbin and GstDiscoverer have seen stability
-    improvements in corner cases such as shutdown while still starting
-    up or shutdown in error cases (hat tip to the oss-fuzz project).
-
--   floating reference handling was inconsistent and has been cleaned up
-    across the board, including annotations. This solves various
-    long-standing memory leaks in language bindings, which e.g. often
-    caused elements and pads to be leaked.
-
--   major gobject-introspection annotation improvements for large parts
-    of the library API, including nullability of return types and
-    function parameters, correct types (e.g. strings vs. filenames),
-    ownership transfer, array length parameters, etc. This allows to use
-    bigger parts of the GStreamer API to be safely used from dynamic
-    language bindings (e.g. Python, Javascript) and allows static
-    bindings (e.g. C#, Rust, Vala) to autogenerate more API bindings
-    without manual intervention.
 
-OpenGL integration
-
--   The GStreamer OpenGL integration library has moved to
-    gst-plugins-base and is now part of our stable API.
+Miscellaneous changes
 
--   new MESA3D GBM BACKEND. On devices with working libdrm support, it
-    is possible to use Mesa3D's GBM library to set up an EGL context
-    directly on top of KMS. This makes it possible to use the GStreamer
-    OpenGL elements without a windowing system if a libdrm- and
-    Mesa3D-supported GPU is present.
+-   this section will be filled in in due course
 
--   Prefer wayland display over X11: As most Wayland compositors support
-    XWayland, the X11 backend would get selected.
+OpenGL integration
 
--   gldownload can export dmabufs now, and glupload will advertise
-    dmabuf as caps feature.
+-   this section will be filled in in due course
 
 
 Tracing framework and debugging improvements
 
--   NEW MEMORY RINGBUFFER BASED DEBUG LOGGER, useful for long-running
-    applications or to retrieve diagnostics when encountering an error.
-    The GStreamer debug logging system provides in-depth debug logging
-    about what is going on inside a pipeline. When enabled, debug logs
-    are usually written into a file, printed to the terminal, or handed
-    off to a log handler installed by the application. However, at
-    higher debug levels the volume of debug output quickly becomes
-    unmanageable, which poses a problem in disk-space or bandwidth
-    restricted environments or with long-running pipelines where a
-    problem might only manifest itself after multiple days. In those
-    situations, developers are usually only interested in the most
-    recent debug log output. The new in-memory ringbuffer logger makes
-    this easy: just installed it with gst_debug_add_ring_buffer_logger()
-    and retrieve logs with gst_debug_ring_buffer_logger_get_logs() when
-    needed. It is possible to limit the memory usage per thread and set
-    a timeout to determine how long messages are kept around. It was
-    always possible to implement this in the application with a custom
-    log handler of course, this just provides this functionality as part
-    of GStreamer.
-
--   'fakevideosink is a null sink for video data that advertises
-    video-specific metas ane behaves like a video sink. See above for
-    more details.
-
--   gst_util_dump_buffer() prints the content of a buffer to stdout.
-
--   gst_pad_link_get_name() and gst_state_change_get_name() print pad
-    link return values and state change transition values as strings.
-
--   The LATENCY TRACER has seen a few improvements: trace records now
-    contain timestamps which is useful to plot things over time, and
-    downstream synchronisation time is now excluded from the measured
-    values.
-
--   Miniobject refcount tracing and logging was not entirley
-    thread-safe, there were duplicates or missing entries at times. This
-    has now been made reliable.
-
--   The netsim element, which can be used to simulate network jitter,
-    packet reordering and packet loss, received new features and
-    improvements: it can now also simulate network congestion using a
-    token bucket algorithm. This can be enabled via the "max-kbps"
-    property. Packet reordering can be disabled now via the
-    "allow-reordering" property: Reordering of packets is not very
-    common in networks, and the delay functions will always introduce
-    reordering if delay > packet-spacing, so by setting
-    "allow-reordering" to FALSE you guarantee that the packets are in
-    order, while at the same time introducing delay/jitter to them. By
-    using the new "delay-distribution" property the user can control how
-    the delay applied to delayed packets is distributed: This is either
-    the uniform distribution (as before) or the normal distribution; in
-    addition there is also the gamma distribution which simulates the
-    delay on wifi networks better.
+-   this section will be filled in in due course
 
 
 Tools
 
--   gst-inspect-1.0 now prints pad properties for elements that have pad
-    subclasses with special properties, such as compositor or
-    audiomixer. This only works for elements that use the newly-added
-    GstPadTemplate API API or the
-    gst_element_class_add_static_pad_template_with_gtype() convenience
-    function to tell GStreamer about the special pad subclass.
-
--   gst-launch-1.0 now generates a gstreamer pipeline diagram (.dot
-    file) whenever SIGHUP is sent to it on Linux/*nix systems.
-
--   gst-discoverer-1.0 can now analyse live streams such as rtsp:// URIs
+-   this section will be filled in in due course
 
 
 GStreamer RTSP server
 
--   Initial support for RTSP protocol version 2.0 was added, which is to
-    the best of our knowledge the first RTSP 2.0 implementation ever!
-
--   ONVIF audio backchannel support. This is an extension specified by
-    ONVIF that allows RTSP clients (e.g. a control room operator) to
-    send audio back to the RTSP server (e.g. an IP camera).
-    Theoretically this could have been done also by using the RECORD
-    method of the RTSP protocol, but ONVIF chose not to do that, so the
-    backchannel is set up alongside the other streams. Format
-    negotiation needs to be done out of band, if needed. Use the new
-    ONVIF-specific subclasses GstRTSPOnvifServer and
-    GstRTSPOnvifMediaFactory to enable this functionality.
-
--   The internal server streaming pipeline is now dynamically
-    reconfigured on PLAY based on the transports needed. This means that
-    the server no longer adds the pipeline plumbing for all possible
-    transports from the start, but only if needed as needed. This
-    improves performance and memory footprint.
-
--   rtspclientsink has gained an "accept-certificate" signal for
-    manually checking a TLS certificate for validity.
-
--   Fix keep-alive/timeout issue for certain clients using TCP
-    interleave as transport who don't do keep-alive via some other
-    method such as periodic RTSP OPTION requests. We now put netaddress
-    metas on the packets from the TCP interleaved stream, so can map
-    RTCP packets to the right stream in the server and can handle them
-    properly.
-
--   Language bindings improvements: in general there were quite a few
-    improvements in the gobject-introspection annotations, but we also
-    extended the permissions API which was not usable from bindings
-    before.
-
--   Fix corner case issue where the wrong mount point was found when
-    there were multiple mount points with a common prefix.
+-   this section will be filled in in due course
 
 
 GStreamer VAAPI
 
--   Improve DMABuf's usage, both upstream and dowstream, and
-    memory:DMABuf caps feature is also negotiated when the dmabuf-based
-    buffer cannot be mapped onto user-space.
-
--   VA initialization was fixed when it is used in headless systems.
-
--   VA display sharing, through GstContext, among the pipeline, has been
-    improved, adding the possibility to the application share its VA
-    display (external display) via gst.vaapi.app.Display context.
-
--   VA display cache was removed.
-
--   libva's log messages are now redirected into the GStreamer log
-    handler.
-
--   Decoders improved their upstream re-negotiation by avoiding to
-    re-instantiate the internal decoder if stream caps are compatible
-    with the previous one.
-
--   When downstream doesn't support GstVideoMeta and the decoded frames
-    don't have standard strides, they are copied onto system
-    memory-based buffers.
-
--   H.264 decoder has a low-latency property, for live streams which
-    doesn't conform the H.264 specification but still it is required to
-    push the frames to downstream as soon as possible.
-
--   As part of the Google Summer of Code 2017 the H.264 decoder drops
-    MVC and SVC frames when base-only property is enabled.
-
--   Added support for libva-2.0 (VA-API 1.0).
-
--   H.264 and H.265 encoders handle Region-Of-Interest metas by adding a
-    delta-qp for every rectangle within the frame specified by those
-    metas.
-
--   Encoders for H.264 and H.265 set the media profile by the downstream
-    caps.
-
--   H.264 encoder inserts an AU delimiter for each encoded frame when
-    aud property is enabled (it is only available for certain drivers
-    and platforms).
-
--   H.264 encoder supports for P and B hierarchical prediction modes.
-
--   All encoders handles a quality-level property, which is a number
-    from 1 to 8, where a lower number means higher quality, but slower
-    processing, and vice-versa.
-
--   VP8 and VP9 encoders support constant bit-rate mode (CBR).
-
--   VP8, VP9 and H.265 encoders support variable bit-rate mode (VBR).
-
--   Resurrected GstGLUploadTextureMeta handling for EGL backends.
-
--   H.265 encoder can configure its number of reference frames via the
-    refs property.
-
--   Add H.264 encoder mbbrc property, which controls the macro-block
-    bitrate as auto, on or off.
-
--   Add H.264 encoder temporal-levels property, to select the number of
-    temporal levels to be included.
-
--   Add to H.264 and H.265 encoders the properties qp-ip and qp-ib, to
-    handle the QP (quality parameter) difference between the I and P
-    frames, and the I and B frames, respectively.
-
--   vaapisink was demoted to marginal rank on Wayland because COGL
-    cannot display YUV surfaces.
+-   this section will be filled in in due course
 
 
 GStreamer Editing Services and NLE
 
--   Handle crossfade in complex scenarios by using the new
-    compositorpad::crossfade-ratio property
-
--   Add API allowing to stop using proxies for clips in the timeline
-
--   Allow management of none square pixel aspect ratios by allowing
-    application to deal with them in the way they want
-
--   Misc fixes around the timeline editing API
+-   this section will be filled in in due course
 
 
 GStreamer validate
 
--   Handle running scenarios on live pipelines (in the "content sense",
-    not the GStreamer one)
-
--   Implement RTSP support with a basic server based on gst-rtsp-server,
-    and add RTSP 1.0 and 2.0 integration tests
-
--   Implement a plugin that allows users to implement configurable
-    tests. It currently can check if a particular element is added a
-    configurable number of time in the pipeline. In the future that
-    plugin should allow us to implement specific tests of any kind in a
-    descriptive way
-
--   Add a verbosity configuration which behaves in a similare way as the
-    gst-launch-1.0 verbose flags allowing the informations to be
-    outputed on any running pipeline when enabling GstValidate.
-
--   Misc optimization in the launcher, making the tests run much faster.
-
-
-GStreamer C# bindings
-
--   Port to the meson build system, autotools support has been removed
-
--   Use a new GlibSharp version, set as a meson subproject
-
--   Update wrapped API to GStreamer 1.14
+-   this section will be filled in in due course
 
--   Removed the need for "glue" code
 
--   Provide a nuget
+GStreamer Python Bindings
 
--   Misc API fixes
+-   this section will be filled in in due course
 
 
 Build and Dependencies
 
--   the new WebRTC support in gst-plugins-bad depends on the GStreamer
-    elements that ship as part of libnice, and libnice version 1.1.14 is
-    required. Also the dtls and srtp plugins.
-
--   gst-plugins-bad no longer depends on the libschroedinger Dirac codec
-    library.
-
--   The srtp plugin can now also be built against libsrtp2.
-
--   some plugins and libraries have moved between modules, see the
-    _Plugin and_ _library moves_ section above, and their respective
-    dependencies have moved with them of course, e.g. the GStreamer
-    OpenGL integration support library and plugin is now in
-    gst-plugins-base, and mpg123, LAME and twoLAME based audio decoder
-    and encoder plugins are now in gst-plugins-good.
-
--   Unify static and dynamic plugin interface and remove plugin specific
-    static build option: Static and dynamic plugins now have the same
-    interface. The standard --enable-static/--enable-shared toggle is
-    sufficient. This allows building static and shared plugins from the
-    same object files, instead of having to build everything twice.
-
--   The default plugin entry point has changed. This will only affect
-    plugins that are recompiled against new GStreamer headers. Binary
-    plugins using the old entry point will continue to work. However,
-    plugins that are recompiled must have matching plugin names in
-    GST_PLUGIN_DEFINE and filenames, as the plugin entry point for
-    shared plugins is now deduced from the plugin filename. This means
-    you can no longer have a plugin called foo living in a file called
-    libfoobar.so or such, the plugin filename needs to match. This might
-    cause problems with some external third party plugin modules when
-    they get rebuilt against GStreamer 1.14.
-
-
-Note to packagers and distributors
-
-A number of libraries, APIs and plugins moved between modules and/or
-libraries in different modules between version 1.12.x and 1.14.x, see
-the _Plugin and_ _library moves_ section above. Some APIs have seen
-minor ABI changes in the course of moving them into the stable APIs
-section.
-
-This means that you should try to ensure that all major GStreamer
-modules are synced to the same major version (1.12 or 1.13/1.14) and can
-only be upgraded in lockstep, so that your users never end up with a mix
-of major versions on their system at the same time, as this may cause
-breakages.
-
-Also, plugins compiled against >= 1.14 headers will not load with
-GStreamer <= 1.12 owing to a new plugin entry point (but plugin binaries
-built against older GStreamer versions will continue to load with newer
-versions of GStreamer of course).
-
-There is also a small structure size related ABI breakage introduced in
-the gst-plugins-bad codecparsers library between version 1.13.90 and
-1.13.91. This should "only" affect gstreamer-vaapi, so anyone who ships
-the release candidates is advised to upgrade those two modules at the
-same time.
+-   this section will be filled in in due course
 
 
 Platform-specific improvements
 
 Android
 
--   ahcsrc (Android camera source) does autofocus now
+-   this section will be filled in in due course
 
 macOS and iOS
 
--   this section will be filled in shortly {FIXME!}
+-   this section will be filled in in due course
 
 Windows
 
--   The GStreamer wasapi plugin was rewritten and should not only be
-    usable now, but in top shape and suitable for low-latency use cases.
-    The Windows Audio Session API (WASAPI) is Microsoft's most modern
-    method for talking with audio devices, and now that the wasapi
-    plugin is up to scratch it is preferred over the directsound plugin.
-    The ranks of the wasapisink and wasapisrc elements have been updated
-    to reflect this. Further improvements include:
-
--   support for more than 2 channels
-
--   a new "low-latency" property to enable low-latency operation (which
-    should always be safe to enable)
-
--   support for the AudioClient3 API which is only available on Windows
-    10: in wasapisink this will be used automatically if available; in
-    wasapisrc it will have to be enabled explicitly via the
-    "use-audioclient3" property, as capturing audio with low latency and
-    without glitches seems to require setting the realtime priority of
-    the entire pipeline to "critical", which cannot be done from inside
-    the element, but has to be done in the application.
-
--   set realtime thread priority to avoid glitches
-
--   allow opening devices in exclusive mode, which provides much lower
-    latency compared to shared mode where WASAPI's engine period is
-    10ms. This can be activated via the "exclusive" property.
-
--   There are now GstDeviceProvider implementations for the wasapi and
-    directsound plugins, so it's now possible to discover both audio
-    sources and audio sinks on Windows via the GstDeviceMonitor API
-
--   debug log timestamps are now higher granularity owing to
-    g_get_monotonic_time() now being used as fallback in
-    gst_utils_get_timestamp(). Before that, there would sometimes be
-    10-20 lines of debug log output sporting the same timestamp.
+-   this section will be filled in in due course
 
 
 Contributors
 
-Aaron Boxer, Adrián Pardini, Adrien SCH, Akinobu Mita, Alban Bedel,
-Alessandro Decina, Alex Ashley, Alicia Boya García, Alistair Buxton,
-Alvaro Margulis, Anders Jonsson, Andreas Frisch, Andrejs Vasiljevs,
-Andrew Bott, Antoine Jacoutot, Antonio Ospite, Antoni Silvestre, Anton
-Obzhirov, Anuj Jaiswal, Arjen Veenhuizen, Arnaud Bonatti, Arun Raghavan,
-Ashish Kumar, Aurélien Zanelli, Ayaka, Branislav Katreniak, Branko
-Subasic, Brion Vibber, Carlos Rafael Giani, Cassandra Rommel, Chris
-Bass, Chris Paulson-Ellis, Christoph Reiter, Claudio Saavedra, Clemens
-Lang, Cyril Lashkevich, Daniel van Vugt, Dave Craig, Dave Johnstone,
-David Evans, David Schleef, Deepak Srivastava, Dimitrios Katsaros,
-Dmitry Zhadinets, Dongil Park, Dustin Spicuzza, Eduard Sinelnikov,
-Edward Hervey, Enrico Jorns, Eunhae Choi, Ezequiel Garcia, fengalin,
-Filippo Argiolas, Florent Thiéry, Florian Zwoch, Francisco Velazquez,
-François Laignel, fvanzile, George Kiagiadakis, Georg Lippitsch, Graham
-Leggett, Guillaume Desmottes, Gurkirpal Singh, Gwang Yoon Hwang, Gwenole
-Beauchesne, Haakon Sporsheim, Haihua Hu, Håvard Graff, Heekyoung Seo,
-Heinrich Fink, Holger Kaelberer, Hoonhee Lee, Hosang Lee, Hyunjun Ko,
-Ian Jamison, James Stevenson, Jan Alexander Steffens (heftig), Jan
-Schmidt, Jason Lin, Jens Georg, Jeremy Hiatt, Jérôme Laheurte, Jimmy
-Ohn, Jochen Henneberg, John Ludwig, John Nikolaides, Jonathan Karlsson,
-Josep Torra, Juan Navarro, Juan Pablo Ugarte, Julien Isorce, Jun Xie,
-Jussi Kukkonen, Justin Kim, Lasse Laursen, Lubosz Sarnecki, Luc
-Deschenaux, Luis de Bethencourt, Marcin Lewandowski, Mario Alfredo
-Carrillo Arevalo, Mark Nauwelaerts, Martin Kelly, Matej Knopp, Mathieu
-Duponchelle, Matteo Valdina, Matt Fischer, Matthew Waters, Matthieu
-Bouron, Matthieu Crapet, Matt Staples, Michael Catanzaro, Michael
-Olbrich, Michael Shigorin, Michael Tretter, Michał Dębski, Michał Górny,
-Michele Dionisio, Miguel París, Mikhail Fludkov, Munez, Nael Ouedraogo,
-Neos3452, Nicholas Panayis, Nick Kallen, Nicola Murino, Nicolas
-Dechesne, Nicolas Dufresne, Nirbheek Chauhan, Ognyan Tonchev, Ole André
-Vadla Ravnås, Oleksij Rempel, Olivier Crête, Omar Akkila, Orestis
-Floros, Patricia Muscalu, Patrick Radizi, Paul Kim, Per-Erik Brodin,
-Peter Seiderer, Philip Craig, Philippe Normand, Philippe Renon, Philipp
-Zabel, Pierre Pouzol, Piotr Drąg, Ponnam Srinivas, Pratheesh Gangadhar,
-Raimo Järvi, Ramprakash Jelari, Ravi Kiran K N, Reynaldo H. Verdejo
-Pinochet, Rico Tzschichholz, Robert Rosengren, Roland Peffer, Руслан
-Ижбулатов, Sam Hurst, Sam Thursfield, Sangkyu Park, Sanjay NM, Satya
-Prakash Gupta, Scott D Phillips, Sean DuBois, Sebastian Cote, Sebastian
-Dröge, Sebastian Rasmussen, Sejun Park, Sergey Borovkov, Seungha Yang,
-Shakin Chou, Shinya Saito, Simon Himmelbauer, Sky Juan, Song Bing,
-Sreerenj Balachandran, Stefan Kost, Stefan Popa, Stefan Sauer, Stian
-Selnes, Thiago Santos, Thibault Saunier, Thijs Vermeir, Tim Allen,
-Tim-Philipp Müller, Ting-Wei Lan, Tomas Rataj, Tom Bailey, Tonu Jaansoo,
-U. Artie Eoff, Umang Jain, Ursula Maplehurst, VaL Doroshchuk, Vasilis
-Liaskovitis, Víctor Manuel Jáquez Leal, vijay, Vincent Penquerc'h,
-Vineeth T M, Vivia Nikolaidou, Wang Xin-yu (王昕宇), Wei Feng, Wim
-Taymans, Wonchul Lee, Xabier Rodriguez Calvar, Xavier Claessens,
-XuGuangxin, Yasushi SHOJI, Yi A Wang, Youness Alaoui,
+-   this section will be filled in in due course
 
 ... and many others who have contributed bug reports, translations, sent
 suggestions or helped testing.
 
 
-Bugs fixed in 1.14
+Bugs fixed in 1.16
+
+-   this section will be filled in in due course
 
-More than 800 bugs have been fixed during the development of 1.14.
+More than XXX bugs have been fixed during the development of 1.16.
 
 This list does not include issues that have been cherry-picked into the
-stable 1.12 branch and fixed there as well, all fixes that ended up in
-the 1.12 branch are also included in 1.14.
+stable 1.16 branch and fixed there as well, all fixes that ended up in
+the 1.16 branch are also included in 1.16.
 
 This list also does not include issues that have been fixed without a
 bug report in bugzilla, so the actual number of fixes is much higher.
 
 
-Stable 1.14 branch
+Stable 1.16 branch
 
-After the 1.14.0 release there will be several 1.14.x bug-fix releases
+After the 1.16.0 release there will be several 1.16.x bug-fix releases
 which will contain bug fixes which have been deemed suitable for a
 stable branch, but no new features or intrusive changes will be added to
-a bug-fix release usually. The 1.14.x bug-fix releases will be made from
-the git 1.14 branch, which is a stable branch.
-
-1.14.0
-
-1.14.0 was released on 19 March 2018.
-
-1.14.1
+a bug-fix release usually. The 1.16.x bug-fix releases will be made from
+the git 1.16 branch, which is a stable branch.
 
-The first 1.14 bug-fix release (1.14.1) is scheduled to be released
-around the end of March or beginning of April.
+1.16.0
 
-This release only contains bugfixes and it should be safe to update from
-1.14.0.
+1.16.0 is scheduled to be released around September 2018.
 
 
 Known Issues
 
--   The webrtcdsp element (which is unrelated to the newly-landed
-    GStreamer webrtc support) is currently not shipped as part of the
+-   The webrtcdsp element is currently not shipped as part of the
     Windows binary packages due to a build system issue.
 
 
-Schedule for 1.16
+Schedule for 1.18
 
 Our next major feature release will be 1.16, and 1.15 will be the
 unstable development version leading up to the stable 1.16 release. The
 development of 1.15/1.16 will happen in the git master branch.
 
 The plan for the 1.16 development cycle is yet to be confirmed, but it
-is expected that feature freeze will be around August 2018 followed by
+is expected that feature freeze will be around August 2017 followed by
 several 1.15 pre-releases and the new 1.16 stable release in September.
 
 1.16 will be backwards-compatible to the stable 1.14, 1.12, 1.10, 1.8,
@@ -1196,8 +189,6 @@ several 1.15 pre-releases and the new 1.16 stable release in September.
 
 ------------------------------------------------------------------------
 
-_These release notes have been prepared by Tim-Philipp Müller with_
-_contributions from Sebastian Dröge, Sreerenj Balachandran, Thibault
-Saunier_ _and Víctor Manuel Jáquez Leal._
+_These release notes have been prepared by Tim-Philipp Müller._
 
 _License: CC BY-SA 4.0_