-GSTREAMER 1.16 RELEASE NOTES
+GSTREAMER 1.18 RELEASE NOTES
-GStreamer 1.16 has not been released yet. It is scheduled for release in
-January/February 2019.
+THESE RELEASE NOTES ARE A PLACEHOLDER, PLEASE BEAR WITH US WHILE WE
+FINISH WRITING UP THE REAL THING.
-1.15.x is the unstable development version that is being developed in
-the git master branch and which will eventually result in 1.16.
+GStreamer 1.18.0 has not yet been released. It is scheduled for release
+in summer 2020 now.
-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.
+1.17.x is the unstable development series that is currently being
+developed in the git master branch and which will eventually result in
+1.18, and 1.17.2 is the current development release in that series.
-See https://gstreamer.freedesktop.org/releases/1.16/ for the latest
+The schedule for the 1.18 development cycle is yet to be confirmed, but
+it is expected that feature freeze will be in June/July 2020, followed
+by several 1.17 pre-releases and then a new 1.18 stable release in
+July/August 2020.
+
+1.18 will be backwards-compatible to the stable 1.16, 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/1.18/ for the latest
version of this document.
-_Last updated: Monday 25 January 2019, 15:00 UTC (log)_
+_Last updated: Wednesday 1 July 2020, 23:50 UTC (log)_
Introduction
Highlights
-- GStreamer WebRTC stack gained support for data channels for
- peer-to-peer communication based on SCTP, BUNDLE support, as well as
- support for multiple TURN servers.
-
-- AV1 video codec support for Matroska and QuickTime/MP4 containers
- and more configuration options and supported input formats for the
- AOMedia AV1 encoder
-
-- Support for Closed Captions and other Ancillary Data in video
-
-- Support for planar (non-interleaved) raw audio
-
-- GstVideoAggregator, compositor and OpenGL mixer elements are now in
- -base
-
-- New alternate fields interlace mode where each buffer carries a
- single field
-
-- WebM and Matroska ContentEncryption support in the Matroska demuxer
-
-- new WebKit WPE-based web browser source element
-
-- Video4Linux: HEVC encoding and decoding, JPEG encoding, and improved
- dmabuf import/export
-
-- Hardware-accelerated Nvidia video decoder gained support for VP8/VP9
- decoding, whilst the encoder gained support for H.265/HEVC encoding.
-
-- Many improvements to the Intel Media SDK based hardware-accelerated
- video decoder and encoder plugin (msdk): dmabuf import/export for
- zero-copy integration with other components; VP9 decoding; 10-bit
- HEVC encoding; video post-processing (vpp) support including
- deinterlacing; and the video decoder now handles dynamic resolution
- changes.
-
-- The ASS/SSA subtitle overlay renderer can now handle multiple
- subtitles that overlap in time and will show them on screen
- simultaneously
-
-- The Meson build is now feature-complete (*) and it is now the
- recommended build system on all platforms. The Autotools build is
- scheduled to be removed in the next cycle.
-
-- The GStreamer Rust bindings and Rust plugins module are now
- officially part of upstream GStreamer.
-
-- Many performance improvements
+- FIXME
Major new features and changes
Noteworthy new API
-- GstAggregator has a new "min-upstream-latency" property that forces
- a minimum aggregate latency for the input branches of an aggregator.
- This is useful for dynamic pipelines where branches with a higher
- latency might be added later after the pipeline is already up and
- running and where a change in the latency would be disruptive. This
- only applies to the case where at least one of the input branches is
- live though, it won’t force the aggregator into live mode in the
- absence of any live inputs.
-
-- GstBaseSink gained a "processing-deadline" property and
- setter/getter API to configure a processing deadline for live
- pipelines. The processing deadline is the acceptable amount of time
- to process the media in a live pipeline before it reaches the sink.
- This is on top of the systemic latency that is normally reported by
- the latency query. This defaults to 20ms and should make pipelines
- such as v4l2src ! xvimagesink not claim that all frames are late in
- the QoS events. Ideally, this should replace the "max-lateness"
- property for most applications.
-
-- RTCP Extended Reports (XR) parsing according to RFC 3611:
- Loss/Duplicate RLE, Packet Receipt Times, Receiver Reference Time,
- Delay since the last Receiver (DLRR), Statistics Summary, and VoIP
- Metrics reports. This only provides the ability to parse such
- packets, generation of XR packets is not supported yet and XR
- packets are not automatically parsed by rtpbin / rtpsession but must
- be actively handled by the application.
-
-- a new mode for interlaced video was added where each buffer carries
- a single field of interlaced video, with buffer flags indicating
- whether the field is the top field or bottom field. Top and bottom
- fields are expected to alternate in this mode. Caps for this
- interlace mode must also carry a format:Interlaced caps feature to
- ensure backwards compatibility.
-
-- The video library has gained support for three new raw pixel
- formats:
-
- - Y410: packed 4:4:4 YUV, 10 bits per channel
- - Y210: packed 4:2:2 YUV, 10 bits per channel
- - NV12_10LE40: fully-packed 10-bit variant of NV12_10LE32,
- i.e. without the padding bits
-
-- GstRTPSourceMeta is a new meta that can be used to transport
- information about the origin of depayloaded or decoded RTP buffers,
- e.g. when mixing audio from multiple sources into a single stream. A
- new "source-info" property on the RTP depayloader base class
- determines whether depayloaders should put this meta on outgoing
- buffers. Similarly, the same property on RTP payloaders determines
- whether they should use the information from this meta to construct
- the CSRCs list on outgoing RTP buffers.
-
-- gst_sdp_message_from_text() is a convenience constructor to parse
- SDPs from a string which is particularly useful for language
- bindings.
-
-Support for Planar (Non-Interleaved) Raw Audio
-
-Raw audio samples are usually passed around in interleaved form in
-GStreamer, which means that if there are multiple audio channels the
-samples for each channel are interleaved in memory, e.g.
-|LEFT|RIGHT|LEFT|RIGHT|LEFT|RIGHT| for stereo audio. A non-interleaved
-or planar arrangement in memory would look like
-|LEFT|LEFT|LEFT|RIGHT|RIGHT|RIGHT| instead, possibly with
-|LEFT|LEFT|LEFT| and |RIGHT|RIGHT|RIGHT| residing in separate memory
-chunks or separated by some padding.
-
-GStreamer has always had signalling for non-interleaved audio since
-version 1.0, but it was never actually properly implemented in any
-elements. audioconvert would advertise support for it, but wasn’t
-actually able to handle it correctly.
-
-With this release we now have full support for non-interleaved audio as
-well, which means more efficient integration with external APIs that
-handle audio this way, but also more efficient processing of certain
-operations like interleaving multiple 1-channel streams into a
-multi-channel stream which can be done without memory copies now.
-
-New API to support this has been added to the GStreamer Audio support
-library: There is now a new GstAudioMeta which describes how data is
-laid out inside the buffer, and buffers with non-interleaved audio must
-always carry this meta. To access the non-interleaved audio samples you
-must map such buffers with gst_audio_buffer_map() which works much like
-gst_buffer_map() or gst_video_frame_map() in that it will populate a
-little GstAudioBuffer helper structure passed to it with the number of
-samples, the number of planes and pointers to the start of each plane in
-memory. This function can also be used to map interleaved audio buffers
-in which case there will be only one plane of interleaved samples.
-
-Of course support for this has also been implemented in the various
-audio helper and conversion APIs, base classes, and in elements such as
-audioconvert, audioresample, audiotestsrc, audiorate.
-
-Support for Closed Captions and Other Ancillary Data in Video
-
-The video support library has gained support for detecting and
-extracting Ancillary Data from videos as per the SMPTE S291M
-specification, including:
-
-- a VBI (Vertical Blanking Interval) parser that can detect and
- extract Ancillary Data from Vertical Blanking Interval lines of
- component signals. This is currently supported for videos in v210
- and UYVY format.
-
-- a new GstMeta for closed captions: GstVideoCaptionMeta. This
- supports the two types of closed captions, CEA-608 and CEA-708,
- along with the four different ways they can be transported (other
- systems are a superset of those).
-
-- a VBI (Vertical Blanking Interval) encoder for writing ancillary
- data to the Vertical Blanking Interval lines of component signals.
-
-The new closedcaption plugin in gst-plugins-bad then makes use of all
-this new infrastructure and provides the following elements:
-
-- cccombiner: a closed caption combiner that takes a closed captions
- stream and another stream and adds the closed captions as
- GstVideoCaptionMeta to the buffers of the other stream.
-
-- ccextractor: a closed caption extractor which will take
- GstVideoCaptionMeta from input buffers and output them as a separate
- closed captions stream.
-
-- ccconverter: a closed caption converter that can convert between
- different formats
-
-- line21decoder: extract line21 closed captions from SD video streams
-
-- cc708overlay: decodes CEA 608/708 captions and overlays them on
- video
-
-Additionally, the following elements have also gained Closed Caption
-support:
-
-- qtdemux and qtmux support CEA 608/708 Closed Caption tracks
-
-- mpegvideoparse extracts Closed Captions from MPEG-2 video streams
-
-- decklinkvideosink can output closed captions and decklinkvideosrc
- can extract closed captions
-
-- playbin and playbin3 learned how to autoplug CEA 608/708 CC overlay
- elements
-
-- the externally maintained ajavideosrc element for AJA capture cards
- has support for extracting closed captions
-
-The rsclosedcaption plugin in the Rust plugins collection includes a
-MacCaption (MCC) file parser and encoder.
+- FIXME
New Elements
-- overlaycomposition: New element that allows applications to draw
- GstVideoOverlayCompositions on a stream. The element will emit the
- "draw" signal for each video buffer, and the application then
- generates an overlay for that frame (or not). This is much more
- performant than e.g. cairooverlay for many use cases, e.g. because
- pixel format conversions can be avoided or the blitting of the
- overlay can be delegated to downstream elements (such as
- gloverlaycompositor). It’s particularly useful for cases where only
- a small section of the video frame should be drawn on.
-
-- gloverlaycompositor: New OpenGL-based compositor element that
- flattens any overlays from GstVideoOverlayCompositionMetas into the
- video stream. This element is also always part of glimagesink.
-
-- glalpha: New element that adds an alpha channel to a video stream.
- The values of the alpha channel can either be set to a constant or
- can be dynamically calculated via chroma keying. It is similar to
- the existing alpha element but based on OpenGL. Calculations are
- done in floating point so results may not be identical to the output
- of the existing alpha element.
-
-- rtpfunnel funnels together RTP streams into a single session. Use
- cases include multiplexing and bundle. webrtcbin uses it to
- implement BUNDLE support.
-
-- testsrcbin is a source element that provides an audio and/or video
- stream and also announces them using the recently-introduced
- GstStream API. This is useful for testing elements such as playbin3
- or uridecodebin3 etc.
-
-- New closed caption elements: cccombiner, ccextractor, ccconverter,
- line21decoder and cc708overlay (see above)
-
-- wpesrc: new source element acting as a Web Browser based on WebKit
- WPE
-
-- Two new OpenCV-based elements: cameracalibrate and cameraundistort
- that can communicate to figure out distortion correction parameters
- for a camera and correct for the distortion.
-
-- New sctp plugin based on usrsctp with sctpenc and sctpdec elements.
- These elements are used inside webrtcbin for implementing data
- channels.
+- FIXME
New element features and additions
-- playbin3, playbin and playsink have gained a new "text-offset"
- property to adjust the positioning of the selected subtitle stream
- vis-a-vis the audio and video streams. This uses subtitleoverlay’s
- new "subtitle-ts-offset" property. GstPlayer has gained matching API
- for this, namely gst_player_get_text_video_offset().
-
-- playbin3 buffering improvements: in network playback scenarios there
- may be multiple inputs to decodebin3, and buffering will be done
- before decodebin3 using queue2 or downloadbuffer elements inside
- urisourcebin. Since this is before any parsers or demuxers there may
- not be any bitrate information available for the various streams, so
- it was difficult to configure the buffering there smartly within
- global constraints. This was improved now: The queue2 elements
- inside urisourcebin will now use the new bitrate query to figure out
- a bitrate estimate for the stream if no bitrate was provided by
- upstream, and urisourcebin will use the bitrates of the individual
- queues to distribute the globally-set "buffer-size" budget in bytes
- to the various queues. urisourcebin also gained "low-watermark" and
- "high-watermark" properties which will be proxied to the internal
- queues, as well as a read-only "statistics" property which allows
- querying of the minimum/maximum/average byte and time levels of the
- queues inside the urisourcebin in question.
-
-- splitmuxsink has gained a couple of new features:
-
- - new "async-finalize" mode: This mode is useful for muxers or
- outputs that can take a long time to finalize a file. Instead of
- blocking the whole upstream pipeline while the muxer is doing
- its stuff, we can unlink it and spawn a new muxer + sink
- combination to continue running normally. This requires us to
- receive the muxer and sink (if needed) as factories via the new
- "muxer-factory" and "sink-factory" properties, optionally
- accompanied by their respective properties structures (set via
- the new "muxer-properties" and "sink-properties" properties).
- There are also new "muxer-added" and "sink-added" signals in
- case custom code has to be called for them to configure them.
-
- - "split-at-running-time" action signal: When called by the user,
- this action signal ends the current file (and starts a new one)
- as soon as the given running time is reached. If called multiple
- times, running times are queued up and processed in the order
- they were given.
-
- - "split-after" action signal to finish outputting the current GOP
- to the current file and then start a new file as soon as the GOP
- is finished and a new GOP is opened (unlike the existing
- "split-now" which immediately finishes the current file and
- writes the current GOP into the next newly-started file).
-
- - "reset-muxer" property: when unset, the muxer is reset using
- flush events instead of setting its state to NULL and back. This
- means the muxer can keep state across resets, e.g. mpegtsmux
- will keep the continuity counter continuous across segments as
- required by hlssink2.
-
-- qtdemux gained PIFF track encryption box support in addition to the
- already-existing PIFF sample encryption support, and also allows
- applications to select which encryption system to use via a
- "drm-preferred-decryption-system-id" context in case there are
- multiple options.
-
-- qtmux: the "start-gap-threshold" property determines now whether an
- edit list will be created to account for small gaps or offsets at
- the beginning of a stream in case the start timestamps of tracks
- don’t line up perfectly. Previously the threshold was hard-coded to
- 1% of the (video) frame duration, now it is 0 by default (so edit
- list will be created even for small differences), but fully
- configurable.
-
-- rtpjitterbuffer has improved end-of-stream handling
-
-- rtpmp4vpay will be prefered over rtpmp4gpay for MPEG-4 video in
- autoplugging scenarios now
-
-- rtspsrc now allows applications to send RTSP SET_PARAMETER and
- GET_PARAMETER requests using action signals.
-
-- rtspsrc has a small (100ms) configurable teardown delay by default
- to try and make sure an RTSP TEARDOWN request gets sent out when the
- source element shuts down. This will block the downward PAUSED to
- READY state change for a short time, but can be disabled where it’s
- a problem. Some servers only allow a limited number of concurrent
- clients, so if no proper TEARDOWN is sent new clients may have
- problems connecting to the server for a while.
-
-- souphttpsrc behaves better with low bitrate streams now. Before it
- would increase the read block size too quickly which could lead to
- it not reading any data from the socket for a very long time with
- low bitrate streams that are output live downstream. This could lead
- to servers kicking off the client.
-
-- filesink: do internal buffering to avoid performance regression with
- small writes since we bypass libc buffering by using writev()
- instead of fwrite()
-
-- identity: add "eos-after" property and fix "error-after" property
- when the element is reused
-
-- input-selector: lets context queries pass through, so that
- e.g. upstream OpenGL elements can use contexts and displays
- advertised by downstream elements
-
-- queue2: avoid ping-pong between 0% and 100% buffering messages if
- upstream is pushing buffers larger than one of its limits, plus
- performance optimisations
-
-- opusdec: new "phase-inversion" property to control phase inversion.
- When enabled, this will slightly increase stereo quality, but
- produces a stream that when downmixed to mono will suffer audio
- distortions.
-
-- The x265enc HEVC encoder also exposes a "key-int-max" property to
- configure the maximum allowed GOP size now.
-
-- decklinkvideosink has seen stability improvements for long-running
- pipelines (potential crash due to overflow of leaked clock refcount)
- and clock-slaving improvements when performing flushing seeks
- (causing stalls in the output timeline), pausing and/or buffering.
-
-- srtpdec, srtpenc: add support for MKIs which allow multiple keys to
- be used with a single SRTP stream
-
-- The srt Secure Reliable Transport plugin has integrated server and
- client elements srt{client,server}{src,sink} into one (srtsrc and
- srtsink), since SRT connection mode can be changed by uri
- parameters.
-
-- h264parse and h265parse will handle SEI recovery point messages and
- mark recovery points as keyframes as well (in addition to IDR
- frames)
-
-- webrtcbin: "add-turn-server" action signal to pass multiple ICE
- relays (TURN servers).
-
-- The removesilence element has received various new features and
- properties, such as a "threshold" property, detecting silence only
- after minimum silence time/buffers, a "silent" property to control
- bus message notifications as well as a "squash" property.
-
-- AOMedia AV1 decoder gained support for 10/12bit decoding whilst the
- AV1 encoder supports more image formats and subsamplings now and
- acquired support for rate control and profile related configuration.
-
-- The Fraunhofer fdkaac plugin can now be built against the 2.0.0
- version API and has improved multichannel support
-
-- kmssink now supports unpadded 24-bit RGB and can configure mode
- setting from video info, which enables display of multi-planar
- formats such as I420 or NV12 with modesetting. It has also gained a
- number of new properties: The "restore-crtc" property does what it
- says on the tin and is enabled by default. "plane-properties" and
- "connector-properties" can be used to pass custom properties to the
- DRM.
-
-- waylandsink has a "fullscreen" property now.
+- FIXME
Plugin and library moves
-- The stereo element was moved from -bad into the existing audiofx
- plugin in -good. If you get duplicate type registration warnings
- when upgrading, check that you don’t have a stale stereoplugin lying
- about somewhere.
-
-GstVideoAggregator, compositor, and OpenGL mixer elements moved from -bad to -base
-
-GstVideoAggregator is a new base class for raw video mixers and muxers
-and is based on GstAggregator. It provides defined-latency mixing of raw
-video 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.
-Most notably, the "ignore-eos" pad property was renamed to
-"repeat-after-eos" and the conversion code was moved to a
-GstVideoAggregatorConvertPad subclass to avoid code duplication, make
-things less awkward for subclasses like the OpenGL-based video mixer,
-and make the API more consistent with the audio aggregator API.
-
-It is used by the compositor element, which is a replacement for
-‘videomixer’ which did not handle live inputs very well. compositor
-should behave much better in that respect and generally behave as one
-would expected in most scenarios.
-
-The compositor element has gained support for per-pad blending mode
-operators (SOURCE, OVER, ADD) which determines what operator to use for
-blending this pad over the previous ones. This can be used to implement
-crossfading and the available operators can be extended in the future as
-needed.
-
-A number of OpenGL-based video mixer elements (glvideomixer, glmixerbin,
-glvideomixerelement, glstereomix, glmosaic) which are built on top of
-GstVideoAggregator have also been moved from -bad to -base now. These
-elements have been merged into the existing OpenGL plugin, so if you get
-duplicate type registration warnings when upgrading, check that you
-don’t have a stale openglmixers plugin lying about somewhere.
+- FIXME
Plugin removals
The following plugins have been removed from gst-plugins-bad:
-- The experimental daala plugin has been removed, since it’s not so
- useful now that all effort is focused on AV1 instead, and it had to
- be enabled explicitly with --enable-experimental anyway.
-
-- The spc plugin has been removed. It has been replaced by the gme
- plugin.
-
-- The acmmp3dec and acmenc plugins for Windows have been removed. ACM
- is an ancient legacy API and there was no point in keeping the
- plugins around for a licensed MP3 decoder now that the MP3 patents
- have expired and we have a decoder in -good. We also didn’t ship
- these in our cerbero-built Windows packages, so it’s unlikely that
- they’ll be missed.
+- FIXME
Miscellaneous API additions
-- GstBitwriter: new generic bit writer API to complement the existing
- bit reader
-
-- gst_buffer_new_wrapped_bytes() creates a wrap buffer from a GBytes
-
-- gst_caps_set_features_simple() sets a caps feature on all the
- structures of a GstCaps
-
-- New GST_QUERY_BITRATE query: This allows determining from downstream
- what the expected bitrate of a stream may be which is useful in
- queue2 for setting time based limits when upstream does not provide
- timing information. tsdemux, qtdemux and matroskademux have basic
- support for this query on their sink pads.
-
-- elements: there is a new “Hardware” class specifier. Elements
- interacting with hardware devices should specify this classifier in
- their element factory class metadata. This is useful to advertise as
- one might need to put such elements into READY state to test if the
- hardware is present in the system for example.
-
-- protection: Add a new definition for unspecified system protection,
- GST_PROTECTION_UNSPECIFIED_SYSTEM_ID
-
-- take functions for various mini objects that didn’t have them yet:
- gst_query_take(), gst_message_take(), gst_tag_list_take(),
- gst_buffer_list_take(). Unlike the various _replace() functions
- _take() does not increase the reference count but takes ownership of
- the mini object passed.
-
-- clear functions for various mini object types and GstObject which
- unrefs the object or mini object (if non-NULL) and sets the variable
- pointed to to NULL: gst_clear_structure(), gst_clear_tag_list(),
- gst_clear_query(), gst_clear_message(), gst_clear_event(),
- gst_clear_caps(), gst_clear_buffer_list(), gst_clear_buffer(),
- gst_clear_mini_object(), gst_clear_object()
-
-- miniobject: new API gst_mini_object_add_parent() and
- gst_mini_object_remove_parent() to set parent pointers on mini
- objects to ensure correct writability: Every container of
- miniobjects now needs to store itself as parent in the child object,
- and remove itself again later. A mini object is then only writable
- if there is at most one parent, that parent is writable itself, and
- the reference count of the mini object is 1. GstBuffer (for
- memories), GstBufferList (for buffers), GstSample (for caps, buffer,
- bufferlist), and GstVideoOverlayComposition were updated
- accordingly. Without this it was possible to have e.g. a buffer list
- with a refcount of 2 used in two places at once that both modify the
- same buffer with refcount 1 at the same time wrongly thinking it is
- writable even though it’s really not.
-
-- poll: add API to watch for POLLPRI and stop treating POLLPRI as a
- read. This is useful to wait for video4linux events which are
- signalled via POLLPRI.
-
-- sample: new API to update the contents of a GstSample and make it
- writable: gst_sample_set_buffer(), gst_sample_set_caps(),
- gst_sample_set_segment(), gst_sample_set_info(), plus
- gst_sample_is_writable() and gst_sample_make_writable(). This makes
- it possible to reuse a sample object and avoid unnecessary memory
- allocations, for example in appsink.
-
-- ClockIDs now keep a weak reference to underlying clock to avoid
- crashes in basesink in corner cases where a clock goes away while
- the ClockID is still in use, plus some new API
- (gst_clock_id_get_clock(), gst_clock_id_uses_clock()) to check the
- clock a ClockID is linked to.
-
-- The GstCheck unit test library gained a
- fail_unless_equals_clocktime() convenience macro as well as some new
- GstHarness API for for proposing meta APIs from the allocation
- query: gst_harness_add_propose_allocation_meta(). ASSERT_CRITICAL()
- checks in unit tests are now skipped if GStreamer was compiled with
- GST_DISABLE_GLIB_CHECKS.
-
-- gst_audio_buffer_truncate() convenience function to truncate a raw
- audio buffer
+- FIXME
Miscellaneous performance and memory optimisations
As always there have been many performance and memory usage improvements
-across all components and modules. Some of them (such as dmabuf
-import/export) have already been mentioned elsewhere so won’t be
-repeated here.
+across all components and modules. Some of them have already been
+mentioned elsewhere so won’t be repeated here.
The following list is only a small snapshot of some of the more
interesting optimisations that haven’t been mentioned in other contexts
yet:
-- The GstVideoEncoder and GstVideoDecoder base classes now release the
- STREAM_LOCK when pushing out buffers, which means (multi-threaded)
- encoders and decoders can now receive and continue to process input
- buffers whilst waiting for downstream elements in the pipeline to
- process the buffer that was pushed out. This increases throughput
- and reduces processing latency, also and especially for
- hardware-accelerated encoder/decoder elements.
-
-- GstQueueArray has seen a few API additions
- (gst_queue_array_peek_nth(), gst_queue_array_set_clear_func(),
- gst_queue_array_clear()) so that it can be used in other places like
- GstAdapter instead of a GList, which reduces allocations and
- improves performance.
-
-- appsink now reuses the sample object in pull_sample() if possible
-
-- rtpsession only starts the RTCP thread when it’s actually needed now
-
-- udpsrc uses a buffer pool now and the GstUdpSrc object structure was
- optimised for better cache performance
+- FIXME
GstPlayer
-- API was added to fine-tune the synchronisation offset between
- subtitles and video
+- FIXME
Miscellaneous changes
-- As a result of moving to newer FFmpeg APIs, encoder and decoder
- elements exposed by the GStreamer FFmpeg wrapper plugin (gst-libav)
- may have seen possibly incompatible changes to property names and/or
- types, and not all properties exposed might be functional. We are
- still reviewing the new properties and aim to minimise breaking
- changes at least for the most commonly-used properties, so please
- report any issues you run into!
+- FIXME
OpenGL integration
-- The OpenGL mixer elements have been moved from -bad to
- gst-plugins-base (see above)
-
-- The Mesa GBM backend now supports headless mode
-
-- gloverlaycompositor: New OpenGL-based compositor element that
- flattens any overlays from GstVideoOverlayCompositionMetas into the
- video stream.
-
-- glalpha: New element that adds an alpha channel to a video stream.
- The values of the alpha channel can either be set to a constant or
- can be dynamically calculated via chroma keying. It is similar to
- the existing alpha element but based on OpenGL. Calculations are
- done in floating point so results may not be identical to the output
- of the existing alpha element.
-
-- glupload: Implement direct dmabuf uploader, the idea being that some
- GPUs (like the Vivante series) can actually perform the YUV->RGB
- conversion internally, so no custom conversion shaders are needed.
- To make use of this feature, we need an additional uploader that can
- import DMABUF FDs and also directly pass the pixel format, relying
- on the GPU to do the conversion.
+- FIXME
Tracing framework and debugging improvements
-- There is now a GDB PRETTY PRINTER FOR VARIOUS GSTREAMER TYPES: For
- GstObject pointers the type and name is added, e.g.
- 0x5555557e4110 [GstDecodeBin|decodebin0]. For GstMiniObject pointers
- the object type is added, e.g. 0x7fffe001fc50 [GstBuffer]. For
- GstClockTime and GstClockTimeDiff the time is also printed in human
- readable form, e.g. 150116219955 [+0:02:30.116219955].
-
-- GDB EXTENSION WITH TWO CUSTOM GDB COMMANDS gst-dot AND gst-print:
-
- - gst-dot creates dot files that a very close to what
- GST_DEBUG_BIN_TO_DOT_FILE() produces, but object properties and
- buffer contents such as codec-data in caps are not available.
-
- - gst-print produces high-level information about a GStreamer
- object. This is currently limited to pads for GstElements and
- events for the pads. The output may look like this:
-
- (gdb) gst-print pad.object.parent
- GstMatroskaDemux (matroskademux0) {
- SinkPad (sink, pull) {
- }
- SrcPad (video_0, push) {
- events:
- stream-start:
- stream-id: 0463ccb080d00b8689bf569a435c4ff84f9ff753545318ae2328ea0763fd0bec/001:1274058367
- caps: video/x-theora
- width: 1920
- height: 800
- pixel-aspect-ratio: 1/1
- framerate: 24/1
- streamheader: < 0x5555557c7d30 [GstBuffer], 0x5555557c7e40 [GstBuffer], 0x7fffe00141d0 [GstBuffer] >
- segment: time
- rate: 1
- tag: global
- container-format: Matroska
- }
- SrcPad (audio_0, push) {
- events:
- stream-start:
- stream-id: 0463ccb080d00b8689bf569a435c4ff84f9ff753545318ae2328ea0763fd0bec/002:1551204875
- caps: audio/mpeg
- mpegversion: 4
- framed: true
- stream-format: raw
- codec_data: 0x7fffe0014500 [GstBuffer]
- level: 2
- base-profile: lc
- profile: lc
- channels: 2
- rate: 44100
- segment: time
- rate: 1
- tag: global
- container-format: Matroska
- tag: stream
- audio-codec: MPEG-4 AAC audio
- language-code: en
- }
- }
-
-- gst_structure_to_string() now serialises the actual value of
- pointers when serialising GstStructures instead of claiming they’re
- NULL. This makes debug logging in various places less confusing,
- because it’s clear now that structure fields actually hold valid
- objects. Such object pointer values will never be deserialised
- however.
+- FIXME
Tools
-- gst-inspect-1.0 has coloured output now and will automatically use a
- pager if the output does not fit on a page. This only works in a
- UNIX environment and if the output is not piped, and on Windows 10
- build 16257 or newer. If you don’t like the colours you can disable
- them by setting the GST_INSPECT_NO_COLORS=1 environment variable or
- passing the --no-color command line option.
+- FIXME
GStreamer RTSP server
-- Improved backlog handling when using TCP interleaved for data
- transport. Before there was a fixed maximum size for backlog
- messages, which was prone to deadlocks and made it difficult to
- control memory usage with the watch backlog. The RTSP server now
- limits queued TCP data messages to one per stream, moving queuing of
- the data into the pipeline and leaving the RTSP connection
- responsive to RTSP messages in both directions, preventing all those
- problems.
-
-- Initial ULP Forward Error Correction support in rtspclientsink and
- for RECORD mode in the server.
-
-- API to explicitly enable retransmission requests (RTX)
-
-- Lots of multicast-related fixes
-
-- rtsp-auth: Add support for parsing .htdigest files
+- FIXME
GStreamer VAAPI
-- this section will be filled in in due course
+- FIXME
GStreamer OMX
-- Add support of NV16 format to video encoders input.
-
-- Video decoders now handle the ALLOCATION query to tell upstream
- about the number of buffers they require. Video encoders will also
- use this query to adjust their number of allocated buffers
- preventing starvation when using dynamic buffer mode.
-
-- The OMX_PERFORMANCE debug category has been renamed to OMX_API_TRACE
- and can now be used to track a widder variety of interactions
- between OMX and GStreamer.
-
-- Video encoders will now detect frame rate only changes and will
- inform OMX about it rather than doing a full format reset.
-
-- Various Zynq UltraScale+ specific improvements:
- - Video encoders are now able to import dmabuf from upstream.
- - Support for HEVC range extension profiles and more AVC profiles.
- - We can now request video encoders to generate an IDR using the
- force key unit event.
+- FIXME
GStreamer Editing Services and NLE
-- this section will be filled in in due course
+- FIXME
GStreamer validate
-- this section will be filled in in due course
+- FIXME
GStreamer Python Bindings
-- add binding for gst_pad_set_caps()
-
-- pygobject dependency requirement was bumped to >= 3.8
-
-- new audiotestsrc, audioplot, and mixer plugin examples, and a
- dynamic pipeline example
+- FIXME
GStreamer C# Bindings
-- bindings for the GstWebRTC library
+- FIXME
GStreamer Rust Bindings
-The GStreamer Rust bindings are now officially part of the GStreamer
-project and are also maintained in the GStreamer GitLab.
-
-The releases will generally not be synchronized with the releases of
-other GStreamer parts due to dependencies on other projects.
-
-Also unlike the other GStreamer libraries, the bindings will not commit
-to full API stability but instead will follow the approach that is
-generally taken by Rust projects, e.g.:
-
-1) 0.12.X will be completely API compatible with all other 0.12.Y
- versions.
-2) 0.12.X+1 will contain bugfixes and compatible new feature additions.
-3) 0.13.0 will _not_ be backwards compatible with 0.12.X but projects
- will be able to stay at 0.12.X without any problems as long as they
- don’t need newer features.
-
-The current stable release is 0.12.2 and the next release series will be
-0.13, probably around March 2019.
-
-At this point the bindings cover most of GStreamer core (except for most
-notably GstAllocator and GstMemory), and most parts of the app, audio,
-base, check, editing-services, gl, net. pbutils, player, rtsp,
-rtsp-server, sdp, video and webrtc libraries.
-
-Also included is support for creating subclasses of the following types
-and writing GStreamer plugins:
-
-- gst::Element
-- gst::Bin and gst::Pipeline
-- gst::URIHandler and gst::ChildProxy
-- gst::Pad, gst::GhostPad
-- gst_base::Aggregator and gst_base::AggregatorPad
-- gst_base::BaseSrc and gst_base::BaseSink
-- gst_base::BaseTransform
-
-Changes to 0.12.X since 0.12.0
-
-Fixed
-
-- PTP clock constructor actually creates a PTP instead of NTP clock
-
-Added
-
-- Bindings for GStreamer Editing Services
-- Bindings for GStreamer Check testing library
-- Bindings for the encoding profile API (encodebin)
-
-- VideoFrame, VideoInfo, AudioInfo, StructureRef implements Send and
- Sync now
-- VideoFrame has a function to get the raw FFI pointer
-- From impls from the Error/Success enums to the combined enums like
- FlowReturn
-- Bin-to-dot file functions were added to the Bin trait
-- gst_base::Adapter implements SendUnique now
-- More complete bindings for the gst_video::VideoOverlay interface,
- especially
- gst_video::is_video_overlay_prepare_window_handle_message()
-
-Changed
-
-- All references were updated from GitHub to freedesktop.org GitLab
-- Fix various links in the README.md
-- Link to the correct location for the documentation
-- Remove GitLab badge as that only works with gitlab.com currently
-
-Changes in git master for 0.13
-
-Fixed
-
-- gst::tag::Album is the album tag now instead of artist sortname
-
-Added
-
-- Subclassing infrastructure was moved directly into the bindings,
- making the gst-plugin crate deprecated. This involves many API
- changes but generally cleans up code and makes it more flexible.
- Take a look at the gst-plugins-rs crate for various examples.
-
-- Bindings for CapsFeatures and Meta
-- Bindings for
- ParentBufferMeta,VideoMetaandVideoOverlayCompositionMeta`
-- Bindings for VideoOverlayComposition and VideoOverlayRectangle
-- Bindings for VideoTimeCode
-
-- UniqueFlowCombiner and UniqueAdapter wrappers that make use of the
- Rust compile-time mutability checks and expose more API in a safe
- way, and as a side-effect implement Sync and Send now
-
-- More complete bindings for Allocation Query
-- pbutils functions for codec descriptions
-- TagList::iter() for iterating over all tags while getting a single
- value per tag. The old ::iter_tag_list() function was renamed to
- ::iter_generic() and still provides access to each value for a tag
-- Bus::iter() and Bus::iter_timed() iterators around the corresponding
- ::pop\*() functions
-
-- serde serialization of Value can also handle Buffer now
-
-- Extensive comments to all examples with explanations
-- Transmuxing example showing how to use typefind, multiqueue and
- dynamic pads
-- basic-tutorial-12 was ported and added
-
-Changed
-
-- Rust 1.31 is the minimum supported Rust version now
-- Update to latest gir code generator and glib bindings
-
-- Functions returning e.g. gst::FlowReturn or other “combined” enums
- were changed to return split enums like
- Result<gst::FlowSuccess, gst::FlowError> to allow usage of the
- standard Rust error handling.
-
-- MiniObject subclasses are now newtype wrappers around the underlying
- GstRc<FooRef> wrapper. This does not change the API in any breaking
- way for the current usages, but allows MiniObjects to also be
- implemented in other crates and makes sure rustdoc places the
- documentation in the right places.
-
-- BinExt extension trait was renamed to GstBinExt to prevent conflicts
- with gtk::Bin if both are imported
-
-- Buffer::from_slice() can’t possible return None
-
-- Various clippy warnings
+- FIXME
GStreamer Rust Plugins
-Like the GStreamer Rust bindings, the Rust plugins are now officially
-part of the GStreamer project and are also maintained in the GStreamer
-GitLab.
-
-In the 0.3.x versions this contained infrastructure for writing
-GStreamer plugins in Rust, and a set of plugins.
+- FIXME
-In git master that infrastructure was moved to the GLib and GStreamer
-bindings directly, together with many other improvements that were made
-possible by this, so the gst-plugins-rs repository only contains
-GStreamer elements now.
-Elements included are:
-
-- Tutorials plugin: identity, rgb2gray and sinesrc with extensive
- comments
-
-- rsaudioecho, a port of the audiofx element
-
-- rsfilesrc, rsfilesink
-
-- rsflvdemux, a FLV demuxer. Not feature-equivalent with flvdemux yet
-
-- threadshare plugin: ts-appsrc, ts-proxysrc/sink, ts-queue, ts-udpsrc
- and ts-tcpclientsrc elements that use a fixed number of threads and
- share them between instances. For more background about these
- elements see Sebastian’s talk “When adding more threads adds more
- problems - Thread-sharing between elements in GStreamer” at the
- GStreamer Conference 2017.
-
-- rshttpsrc, a HTTP source around the hyper/reqwest Rust libraries.
- Not feature-equivalent with souphttpsrc yet.
-
-- togglerecord, an element that allows to start/stop recording at any
- time and keeps all audio/video streams in sync.
-
-- mccparse and mccenc, parsers and encoders for the MCC closed caption
- file format.
-
-Changes to 0.3.X since 0.3.0
-
-- All references were updated from GitHub to freedesktop.org GitLab
-- Fix various links in the README.md
-- Link to the correct location for the documentation
-
-Changes in git master for 0.4
-
-- togglerecord: Switch to parking_lot crate for mutexes/condition
- variables for lower overhead
-- Merge threadshare plugin here
-- New closedcaption plugin with mccparse and mccenc elements
-- New identity element for the tutorials plugin
-
-- Register plugins statically in tests instead of relying on the
- plugin loader to find the shared library in a specific place
+Build and Dependencies
-- Update to the latest API changes in the GLib and GStreamer bindings
-- Update to the latest versions of all crates
+- The Autotools build system has finally been removed in favour of the
+ Meson build system. Developers who currently use gst-uninstalled
+ should move to gst-build.
+- API and plugin documentation are no longer built with gtk_doc. The
+ gtk_doc documentation has been removed in favour of a new unified
+ documentation module built with hotdoc. The intention is to
+ distribute the generated documentation in form of tarballs alongside
+ releases.
-Build and Dependencies
-
-- The MESON BUILD SYSTEM BUILD IS NOW FEATURE-COMPLETE (*) and it is
- now the recommended build system on all platforms and also used by
- Cerbero to build GStreamer on all platforms. The Autotools build is
- scheduled to be removed in the next cycle. Developers who currently
- use gst-uninstalled should move to gst-build. The build option
- naming has been cleaned up and made consistent and there are now
- feature options to enable/disable plugins and various other features
- on a case-by-case basis. (*) with the exception of plugin docs which
- will be handled differently in future
-
-- Symbol export in libraries is now controlled via explicit exports
- using symbol visibility or export defines where supported, to ensure
- consistency across all platforms. This also allows libraries to have
- exports that vary based on detected platform features and configure
- options as is the case with the GStreamer OpenGL integration library
- for example. A few symbols that had been exported by accident in
- earlier versions may no longer be exported. These symbols will not
- have had declarations in any public header files then though and
- would not have been usable.
-
-- The GStreamer FFmpeg wrapper plugin (gst-libav) now depends on
- FFmpeg 4.x and uses the new FFmpeg 4.x API and stopped relying on
- ancient API that was removed with the FFmpeg 4.x release. This means
- that it is no longer possible to build this module against an older
- system-provided FFmpeg 3.x version. Use the internal FFmpeg 4.x copy
- instead if you build using autotools, or use gst-libav 1.14.x
- instead which targets the FFmpeg 3.x API and _should_ work fine in
- combination with a newer GStreamer. It’s difficult for us to support
- both old and new FFmpeg APIs at the same time, apologies for any
- inconvenience caused.
-
-- Hardware-accelerated Nvidia video encoder/decoder plugins nvdec and
- nvenc can be built against CUDA Toolkit versions 9 and 10.0 now. The
- dynlink interface has been dropped since it’s deprecated in 10.0.
-
-- The (optional) OpenCV requirement has been bumped to >= 3.0.0 and
- the plugin can also be built against OpenCV 4.x now.
-
-- New sctp plugin based on usrsctp (for WebRTC data channels)
+- FIXME
Cerbero
Cerbero has seen a number of improvements:
-- Cerbero has been ported to Python 3 and requires Python 3.5 or newer
- now
-
-- Source tarballs are now protected by checksums in the recipes to
- guard against download errors and malicious takeover of projects or
- websites. In addition, downloads are only allowed via secure
- transports now and plain HTTP, FTP and git:// transports are not
- allowed anymore.
-
-- There is now a new fetch-bootstrap command which downloads sources
- required for bootstrapping, with an optional --build-tools-only
- argument to match the bootstrap --build-tools-only command.
-
-- The bootstrap, build, package and bundle-source commands gained a
- new --offline switch that ensures that only sources from the cache
- are used and never downloaded via the network. This is useful in
- combination with the fetch and fetch-bootstrap commands that acquire
- sources ahead of time before any build steps are executed. This
- allows more control over the sources used and when sources are
- updated, and is particularly useful for build environments that
- don’t have network access.
-
-- bootstrap --assume-yes will automatically say ‘yes’ to any
- interactive prompts during the bootstrap stage, such as those from
- apt-get or yum.
-
-- bootstrap --system-only will only bootstrap the system without build
- tools.
-
-- Manifest support: The build manifest can be used in continuous
- integration (CI) systems to fixate the Git revision of certain
- projects so that all builds of a pipeline are on the same reference.
- This is used in GStreamer’s gitlab CI for example. It can also be
- used in order to re-produce a specific build. To set a manifest, you
- can set manifest = 'my_manifest.xml' in your configuration file, or
- use the --manifest command line option. The command line option will
- take precendence over anything specific in the configuration file.
-
-- The new build-deps command can be used to build only the
- dependencies of a recipe, without the recipe itself.
-
-- new --list-variants command to list available variants
-
-- variants can now be set on the command line via the -v option as a
- comma-separated list. This overrides any variants set in any
- configuration files.
-
-- new qt5, intelmsdk and nvidia variants for enabling Qt5 and hardware
- codec support. See the Enabling Optional Features with Variants
- section in the Cerbero documentation for more details how to enable
- and use these variants.
-
-- A new -t / --timestamp command line switch makes commands print
- timestamps
+- FIXME
Platform-specific changes and improvements
Android
-- toolchain: update compiler to clang and NDKr18. NDK r18 removed the
- armv5 target and only has Android platforms that target at least
- armv7 so the armv5 target is not useful anymore.
-
-- The way that GIO modules are named has changed due to upstream GLib
- natively adding support for loading static GIO modules. This means
- that any GStreamer application using gnutls for SSL/TLS on the
- Android or iOS platforms (or any other setup using static libraries)
- will fail to link looking for the g_io_module_gnutls_load_static()
- function. The new function name is now
- g_io_gnutls_load(gpointer data). data can be NULL for a static
- library. Look at this commit for the necessary change in the
- examples.
-
-- various build issues on Android have been fixed.
+- FIXME
macOS and iOS
-- various build issues on iOS have been fixed.
-
-- the minimum required iOS version is now 9.0. The difference in
- adoption between 8.0 and 9.0 is 0.1% and the bump to 9.0 fixes some
- build issues.
-
-- The way that GIO modules are named has changed due to upstream GLib
- natively adding support for loading static GIO modules. This means
- that any GStreamer application using gnutls for SSL/TLS on the
- Android or iOS platforms (or any other setup using static libraries)
- will fail to link looking for the g_io_module_gnutls_load_static()
- function. The new function name is now
- g_io_gnutls_load(gpointer data). data can be NULL for a static
- library. Look at this commit for the necessary change in the
- examples.
+- FIXME
Windows
-- The webrtcdsp element is shipped again as part of the Windows binary
- packages, the build system issue has been resolved.
-
-- ‘Inconsistent DLL linkage’ warnings when building with MSVC have
- been fixed
-
-- Hardware-accelerated Nvidia video encoder/decoder plugins nvdec and
- nvenc build on Windows now, also with MSVC and using Meson.
-
-- The ksvideosrc camera capture plugin supports 16-bit grayscale video
- now
+- toolchain upgrade
-- The wasapisrc audio capture element implements loopback recording
- from another output device or sink
-
-- wasapisink recover from low buffer levels in shared mode and some
- exclusive mode fixes
-
-- dshowsrc now implements the GstDeviceMonitor interface
+- FIXME
Contributors
-Aleix Conchillo Flaqué, Alessandro Decina, Alexandru Băluț, Alex Ashley,
-Alexey Chernov, Alicia Boya García, Amit Pandya, Andoni Morales
-Alastruey, Andreas Frisch, Andre McCurdy, Andy Green, Anthony Violo,
-Antoine Jacoutot, Antonio Ospite, Arun Raghavan, Aurelien Jarno,
-Aurélien Zanelli, ayaka, Bananahemic, Bastian Köcher, Branko Subasic,
-Brendan Shanks, Carlos Rafael Giani, Christoph Reiter, Corentin Noël,
-Daeseok Youn, Daniel Drake, Daniel Klamt, Dardo D Kleiner, David Ing,
-David Svensson Fors, Devarsh Thakkar, Dimitrios Katsaros, Edward Hervey,
-Emilio Pozuelo Monfort, Enrique Ocaña González, Ezequiel Garcia, Fabien
-Dessenne, Fabrizio Gennari, Florent Thiéry, Francisco Velazquez,
-Freyr666, Garima Gaur, Gary Bisson, George Kiagiadakis, Georg Lippitsch,
-Georg Ottinger, Geunsik Lim, Göran Jönsson, Guillaume Desmottes, H1Gdev,
-Haihao Xiang, Haihua Hu, Harshad Khedkar, Havard Graff, He Junyan,
-Hoonhee Lee, Hosang Lee, Hyunjun Ko, Ingo Randolf, Iñigo Huguet, James
-Stevenson, Jan Alexander Steffens, Jan Schmidt, Jerome Laheurte, Jimmy
-Ohn, Joakim Johansson, Jochen Henneberg, Johan Bjäreholt, John-Mark
-Bell, John Nikolaides, Jonathan Karlsson, Jonny Lamb, Jordan Petridis,
-Josep Torra, Joshua M. Doe, Jos van Egmond, Juan Navarro, Jun Xie,
-Junyan He, Justin Kim, Kai Kang, Kim Tae Soo, Kirill Marinushkin, Kyrylo
-Polezhaiev, Lars Petter Endresen, Linus Svensson, Louis-Francis
-Ratté-Boulianne, Luis de Bethencourt, Luz Paz, Lyon Wang, Maciej Wolny,
-Marc-André Lureau, Marc Leeman, Marcos Kintschner, Marian Mihailescu,
-Marinus Schraal, Mark Nauwelaerts, Marouen Ghodhbane, Martin Kelly,
-Matej Knopp, Mathieu Duponchelle, Matteo Valdina, Matthew Waters,
-Matthias Fend, memeka, Michael Drake, Michael Gruner, Michael Olbrich,
-Michael Tretter, Miguel Paris, Mike Wey, Mikhail Fludkov, Naveen
-Cherukuri, Nicola Murino, Nicolas Dufresne, Niels De Graef, Nirbheek
-Chauhan, Norbert Wesp, Ognyan Tonchev, Olivier Crête, Omar Akkila,
-Patricia Muscalu, Patrick Radizi, Patrik Nilsson, Paul Kocialkowski, Per
-Forlin, Peter Körner, Peter Seiderer, Petr Kulhavy, Philippe Normand,
-Philippe Renon, Philipp Zabel, Pierre Labastie, Roland Jon, Roman
-Sivriver, Rosen Penev, Russel Winder, Sam Gigliotti, Sean-Der, Sebastian
-Dröge, Seungha Yang, Sjoerd Simons, Snir Sheriber, Song Bing, Soon,
-Thean Siew, Sreerenj Balachandran, Stefan Ringel, Stephane Cerveau,
-Stian Selnes, Suhas Nayak, Takeshi Sato, Thiago Santos, Thibault
-Saunier, Thomas Bluemel, Tianhao Liu, Tim-Philipp Müller, Tomasz
-Andrzejak, Tomislav Tustonić, U. Artie Eoff, Ulf Olsson, Varunkumar
-Allagadapa, Víctor Guzmán, Víctor Manuel Jáquez Leal, Vincenzo Bono,
-Vineeth T M, Vivia Nikolaidou, Wang Fei, wangzq, Whoopie, Wim Taymans,
-Wind Yuan, Wonchul Lee, Xabier Rodriguez Calvar, Xavier Claessens,
-Haihao Xiang, Yacine Bandou, Yeongjin Jeong, Yuji Kuwabara, Zeeshan Ali,
+- FIXME
… and many others who have contributed bug reports, translations, sent
suggestions or helped testing.
-Bugs fixed in 1.16
-
-- this section will be filled in in due course
-
-More than XXX bugs have been fixed during the development of 1.16.
+Stable 1.18 branch
-This list does not include issues that have been cherry-picked into the
-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 1.16 branch
-
-After the 1.16.0 release there will be several 1.16.x bug-fix releases
+After the 1.18.0 release there will be several 1.18.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 1.16.x bug-fix releases will be made from
-the git 1.16 branch, which is a stable branch.
+a bug-fix release usually. The 1.18.x bug-fix releases will be made from
+the git 1.18 branch, which will be a stable branch.
-1.16.0
+1.18.0
-1.16.0 is scheduled to be released around January/February 2019.
+1.18.0 has not been released yet.
Known Issues
-- possibly breaking/incompatible changes to properties of wrapped
- FFmpeg decoders and encoders (see above).
+- FIXME
-- The way that GIO modules are named has changed due to upstream GLib
- natively adding support for loading static GIO modules. This means
- that any GStreamer application using gnutls for SSL/TLS on the
- Android or iOS platforms (or any other setup using static libraries)
- will fail to link looking for the g_io_module_gnutls_load_static()
- function. The new function name is now
- g_io_gnutls_load(gpointer data). See Android/iOS sections above for
- further details.
+Schedule for 1.20
-Schedule for 1.18
+Our next major feature release will be 1.20, and 1.19 will be the
+unstable development version leading up to the stable 1.20 release. The
+development of 1.19/1.20 will happen in the git master branch.
-Our next major feature release will be 1.18, and 1.17 will be the
-unstable development version leading up to the stable 1.18 release. The
-development of 1.17/1.18 will happen in the git master branch.
+The plan for the 1.20 development cycle is yet to be confirmed.
-The plan for the 1.18 development cycle is yet to be confirmed, but it
-is expected that feature freeze will be around July 2019 followed by
-several 1.17 pre-releases and the new 1.18 stable release in
-August/September.
-
-1.18 will be backwards-compatible to the stable 1.16, 1.14, 1.12, 1.10,
-1.8, 1.6, 1.4, 1.2 and 1.0 release series.
+1.20 will be backwards-compatible to the stable 1.18, 1.16, 1.14, 1.12,
+1.10, 1.8, 1.6, 1.4, 1.2 and 1.0 release series.
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_These release notes have been prepared by Tim-Philipp Müller with_
-_contributions from Sebastian Dröge._
+_contributions from … (FIXME)_
_License: CC BY-SA 4.0_