-# GStreamer 1.12 Release Notes
-GStreamer 1.12.0 was originally released on 4th May 2017.
-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 RELEASE NOTES
-As always, this release is again packed with new features, bug fixes and other
-improvements.
-See [https://gstreamer.freedesktop.org/releases/1.12/][latest] for the latest
+GStreamer 1.16 has not been released yet. It is scheduled for release in
+March 2019.
+
+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.
+
+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/1.16/ for the latest
version of this document.
-*Last updated: Thursday 4 May 2017, 11:00 UTC [(log)][gitlog]*
+_Last updated: Wednesday 27 January 2019, 00:30 UTC (log)_
+
+
+Introduction
+
+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!
+
+As always, this release is again packed with many new features, bug
+fixes and other improvements.
+
+
+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
-[latest]: https://gstreamer.freedesktop.org/releases/1.12/
-[gitlog]: https://cgit.freedesktop.org/gstreamer/www/log/src/htdocs/releases/1.12/release-notes-1.12.md
+- New alternate fields interlace mode where each buffer carries a
+ single field
-## Introduction
+- WebM and Matroska ContentEncryption support in the Matroska demuxer
-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!
+- new WebKit WPE-based web browser source element
-As always, this release is again packed with new features, bug fixes and other
-improvements.
+- Video4Linux: HEVC encoding and decoding, JPEG encoding, and improved
+ dmabuf import/export
-## Highlights
+- Hardware-accelerated Nvidia video decoder gained support for VP8/VP9
+ decoding, whilst the encoder gained support for H.265/HEVC encoding.
-- new `msdk` plugin for Intel's Media SDK for hardware-accelerated video
- encoding and decoding on Intel graphics hardware on Windows or Linux.
+- 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.
-- `x264enc` can now use multiple x264 library versions compiled for different
- bit depths at runtime, to transparently provide support for multiple bit
- depths.
+- The ASS/SSA subtitle overlay renderer can now handle multiple
+ subtitles that overlap in time and will show them on screen
+ simultaneously
-- `videoscale` and `videoconvert` now support multi-threaded scaling and
- conversion, which is particularly useful with higher resolution video.
+- 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.
-- `h264parse` will now automatically insert AU delimiters if needed when
- outputting byte-stream format, which improves standard compliance and
- is needed in particular for HLS playback on iOS/macOS.
+- The GStreamer Rust bindings and Rust plugins module are now
+ officially part of upstream GStreamer.
-- `rtpbin` has acquired bundle support for incoming streams
+- Many performance improvements
-## Major new features and changes
-### Noteworthy new API
+Major new features and changes
-- The video library gained support for a number of new video formats:
+Noteworthy new API
- - `GBR_12LE`, `GBR_12BE`, `GBRA_12LE`, `GBRA_12BE` (planar 4:4:4 RGB/RGBA, 12 bits per channel)
- - `GBRA_10LE`, `GBRA_10BE` (planar 4:4:4:4 RGBA, 10 bits per channel)
- - `GBRA` (planar 4:4:4:4 ARGB, 8 bits per channel)
- - `I420_12BE`, `I420_12LE` (planar 4:2:0 YUV, 12 bits per channel)
- - `I422_12BE`,`I422_12LE` (planar 4:2:2 YUV, 12 bits per channel)
- - `Y444_12BE`, `Y444_12LE` (planar 4:4:4 YUV, 12 bits per channel)
- - `VYUY` (another packed 4:2:2 YUV format)
+- 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.
-- The high-level `GstPlayer` API was extended with functions for taking video
- snapshots and enabling accurate seeking. It can optionally also use the
- still-experimental `playbin3` element now.
+- 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.
-### New Elements
+- 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.
-- msdk: new plugin for Intel's Media SDK for hardware-accelerated video encoding
- and decoding on Intel graphics hardware on Windows or Linux. This includes
- an H.264 encoder/decoder (`msdkh264dec`, `msdkh264enc`),
- an H.265 encoder/decoder (`msdkh265dec`, `msdkh265enc`),
- an MJPEG encoder/encoder (`msdkmjpegdec`, `msdkmjpegenc`),
- an MPEG-2 video encoder (`msdkmpeg2enc`) and a VP8 encoder (`msdkvp8enc`).
+- 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.
-- `iqa` is a new Image Quality Assessment plugin based on [DSSIM][dssim],
- similar to the old (unported) videomeasure element.
+- The video library has gained support for three new raw pixel
+ formats:
-- The `faceoverlay` element, which allows you to overlay SVG graphics over
- a detected face in a video stream, has been ported from 0.10.
+ - 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
-- our `ffmpeg` wrapper plugin now exposes/maps the ffmpeg Opus audio decoder
- (`avdec_opus`) as well as the GoPro CineForm HD / CFHD decoder (`avdec_cfhd`),
- and also a parser/writer for the IVF format (`avdemux_ivf` and `avmux_ivf`).
+- 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.
-- `audiobuffersplit` is a new element that splits raw audio buffers into
- equal-sized buffers
+- gst_sdp_message_from_text() is a convenience constructor to parse
+ SDPs from a string which is particularly useful for language
+ bindings.
-- `audiomixmatrix` is a new element that mixes N:M audio channels according to
- a configured mix matrix.
+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.
+
+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.
+
+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.
+
+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.
+
+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.
+
+
+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
+
+
+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.
+
+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
+
+GstPlayer
+
+- API was added to fine-tune the synchronisation offset between
+ subtitles and video
+
+
+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!
+
+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.
+
+
+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
+ }
+ }
-- The `timecodewait` element got renamed to `avwait` and can operate in
- different modes now.
+- 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.
-- The `opencv` video processing plugin has gained a new `dewarp` element that
- dewarps fisheye images.
-- `ttml` is a new plugin for parsing and rendering subtitles in Timed Text
- Markup Language (TTML) format. For the time being these elements will not
- be autoplugged during media playback however, unless the `GST_TTML_AUTOPLUG=1`
- environment variable is set. Only the EBU-TT-D profile is supported at this
- point.
+Tools
-[dssim]: https://github.com/pornel/dssim
+- 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.
-### New element features and additions
-- `x264enc` can now use multiple x264 library versions compiled for different
- bit depths at runtime, to transparently provide support for multiple bit
- depths. A new configure parameter `--with-x264-libraries` has been added to
- specify additional paths to look for additional x264 libraries to load.
- Background is that the libx264 library is always compile for one specific
- bit depth and the `x264enc` element would simply support the depth supported
- by the underlying library. Now we can support multiple depths.
+GStreamer RTSP server
-- `x264enc` also picks up the interlacing mode automatically from the input
- caps now and passed interlacing/TFF information correctly to the library.
+- 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.
-- `videoscale` and `videoconvert` now support multi-threaded scaling and
- conversion, which is particularly useful with higher resolution video.
- This has to be enabled explicitly via the `"n-threads"` property.
+- Initial ULP Forward Error Correction support in rtspclientsink and
+ for RECORD mode in the server.
-- `videorate`'s new `"rate"` property lets you set a speed factor
- on the output stream
+- API to explicitly enable retransmission requests (RTX)
-- `splitmuxsink`'s buffer collection and scheduling was rewritten to make
- processing and splitting deterministic; before it was possible for a buffer
- to end up in a different file chunk in different runs. `splitmuxsink` also
- gained a new `"format-location-full"` signal that works just like the existing
- `"format-location"` signal only that it is also passed the primary stream's
- first buffer as argument, so that it is possible to construct the file name
- based on metadata such as the buffer timestamp or any GstMeta attached to
- the buffer. The new `"max-size-timecode"` property allows for timecode-based
- splitting. `splitmuxsink` will now also automatically start a new file if the
- input caps change in an incompatible way.
+- Lots of multicast-related fixes
-- `fakesink` has a new `"drop-out-of-segment"` property to not drop
- out-of-segment buffers, which is useful for debugging purposes.
+- rtsp-auth: Add support for parsing .htdigest files
-- `identity` gained a `"ts-offset"` property.
-- both `fakesink` and `identity` now also print what kind of metas are attached
- to buffers when printing buffer details via the `"last-message"` property
- used by `gst-launch-1.0 -v`.
+GStreamer VAAPI
-- multiqueue: made `"min-interleave-time"` a configurable property.
+- this section will be filled in in due course
-- video nerds will be thrilled to know that `videotestsrc`'s snow is now
- deterministic. `videotestsrc` also gained some new properties to make the
- ball pattern based on system time, and invert colours each second
- (`"animation-mode"`, `"motion"`, and `"flip"` properties).
-
-- `oggdemux` reverse playback should work again now. You're welcome.
-
-- `playbin3` and `urisourcebin` now have buffering enabled by default, and
- buffering message aggregation was fixed.
-
-- `tcpclientsrc` now has a `"timeout"` property
-
-- `appsink` has gained support for buffer lists. For backwards compatibility
- reasons users need to enable this explicitly with `gst_app_sink_set_buffer_list_support()`,
- however. Once activated, a pulled `GstSample` can contain either a buffer
- list or a single buffer.
-
-- `splitmuxsrc` reverse playback was fixed and handling of sparse streams, such
- as subtitle tracks or metadata tracks, was improved.
-
-- `matroskamux` has acquired support for muxing G722 audio; it also marks all
- buffers as keyframes now when streaming only audio, so that `tcpserversink`
- will behave properly with audio-only streams.
-
-- `qtmux` gained support for ProRes 4444 XQ, HEVC/H.265 and CineForm (GoPro) formats,
- and generally writes more video stream-related metadata into the track headers.
- It is also allows configuration of the maximum interleave size in bytes and
- time now. For fragmented mp4 we always write the `tfdt` atom now as required
- by the DASH spec.
-
-- `qtdemux` supports FLAC, xvid, mp2, S16L and CineForm (GoPro) tracks now, and
- generally tries harder to extract more video-related information from track
- headers, such as colorimetry or interlacing details. It also received a
- couple of fixes for the scenario where upstream operates in TIME format and
- feeds chunks to qtdemux (e.g. DASH or MSE).
-
-- `audioecho` has two new properties to apply a delay only to certain channels
- to create a surround effect, rather than an echo on all channels. This is
- useful when upmixing from stereo, for example. The `"surround-delay"` property
- enables this, and the `"surround-mask"` property controls which channels
- are considered surround sound channels in this case.
-
-- `webrtcdsp` gained various new properties for gain control and also exposes
- voice activity detection now, in which case it will post `"voice-activity"`
- messages on the bus whenever the voice detection status changes.
-
-- The `decklink` capture elements for Blackmagic Decklink cards have seen a
- number of improvements:
-
- - `decklinkvideosrc` will post a warning message on "no signal" and an info
- message when the signal lock has been (re)acquired. There is also a new
- read-only `"signal"` property that can be used to query the signal lock
- status. The `GAP` flag will be set on buffers that are captured without
- a signal lock. The new `drop-no-signal-frames` will make `decklinkvideosrc`
- drop all buffers that have been captured without an input signal. The
- `"skip-first-time"` property will make the source drop the first few
- buffers, which is handy since some devices will at first output buffers
- with the wrong resolution before they manage to figure out the right input
- format and decide on the actual output caps.
-
- - `decklinkaudiosrc` supports more than just 2 audio channels now.
-
- - The capture sources no longer use the "hardware" timestamps which turn
- out to be useless and instead just use the pipeline clock directly.
-
-- `srtpdec` now also has a readonly `"stats"` property, just like `srtpenc`.
-
-- `rtpbin` gained RTP bundle support, as used by e.g. WebRTC. The first
- rtpsession will have a `rtpssrcdemux` element inside splitting the streams
- based on their SSRC and potentially dispatch to a different rtpsession.
- Because retransmission SSRCs need to be merged with the corresponding media
- stream the `::on-bundled-ssrc` signal is emitted on `rtpbin` so that the
- application can find out to which session the SSRC belongs.
-
-- `rtprtxqueue` gained two new properties exposing retransmission
- statistics (`"requests"` and `"fulfilled-requests"`)
-
-- `kmssink` will now use the preferred mode for the monitor and render to the
- base plane if nothing else has set a mode yet. This can also be done forcibly
- in any case via the new `"force-modesetting"` property. Furthermore, `kmssink`
- now allows only the supported connector resolutions as input caps in order to
- avoid scaling or positioning of the input stream, as `kmssink` can't know
- whether scaling or positioning would be more appropriate for the use case at
- hand.
-
-- `waylandsink` can now take DMAbuf buffers as input in the presence
- of a compatible Wayland compositor. This enables zero-copy transfer
- from a decoder or source that outputs DMAbuf.
-
-- `udpsrc` can be bound to more than one interface when joining a
- multicast group, this is done by giving a comma separate list of
- interfaces such as multicast-iface="eth0,eth1".
-
-### Plugin moves
-
-- `dataurisrc` moved from gst-plugins-bad to core
-
-- The `rawparse` plugin containing the `rawaudioparse` and `rawvideoparse`
- elements moved from gst-plugins-bad to gst-plugins-base. These elements
- supersede the old `videoparse` and `audioparse` elements. They work the
- same, with just some minor API changes. The old legacy elements still
- exist in gst-plugins-bad, but may be removed at some point in the future.
-
-- `timecodestamper` is an element that attaches time codes to video buffers
- in form of `GstVideoTimeCodeMeta`s. It had a `"clock-source"` property
- which has now been removed because it was fairly useless in practice. It
- gained some new properties however: the `"first-timecode"` property can
- be used to set the inital timecode; alternatively `"first-timecode-to-now"`
- can be set, and then the current system time at the time the first buffer
- arrives is used as base time for the time codes.
-
-
-### Plugin removals
-
-- The `mad` mp1/mp2/mp3 decoder plugin was removed from gst-plugins-ugly,
- as libmad is GPL licensed, has been unmaintained for a very long time, and
- there are better alternatives available. Use the `mpg123audiodec` element
- from the `mpg123` plugin in gst-plugins-ugly instead, or `avdec_mp3` from
- the `gst-libav` module which wraps the ffmpeg library. We expect that we
- will be able to move mp3 decoding to gst-plugins-good in the next cycle
- seeing that most patents around mp3 have expired recently or are about to
- expire.
-
-- The `mimic` plugin was removed from gst-plugins-bad. It contained a decoder
- and encoder for a video codec used by MSN messenger many many years ago (in
- a galaxy far far away). The underlying library is unmaintained and no one
- really needs to use this codec any more. Recorded videos can still be played
- back with the MIMIC decoder in gst-libav.
-
-## Miscellaneous API additions
-
-- Request pad name templates passed to `gst_element_request_pad()` may now
- contain multiple specifiers, such as e.g. `src_%u_%u`.
-
-- [`gst_buffer_iterate_meta_filtered()`][buffer-iterate-meta-filtered] is a
- variant of `gst_buffer_iterate_meta()` that only returns metas of the
- requested type and skips all other metas.
-
-- [`gst_pad_task_get_state()`][pad-task-get-state] gets the current state of
- a task in a thread-safe way.
-
-- [`gst_uri_get_media_fragment_table()`][uri-get-fragment-table] provides the
- media fragments of an URI as a table of key=value pairs.
-
-- [`gst_print()`][print], [`gst_println()`][println], [`gst_printerr()`][printerr],
- and [`gst_printerrln()`][printerrln] can be used to print to stdout or stderr.
- These functions are similar to `g_print()` and `g_printerr()` but they also
- support all the additional format specifiers provided by the GStreamer
- logging system, such as e.g. `GST_PTR_FORMAT`.
-
-- a `GstParamSpecArray` has been added, for elements who want to have array
- type properties, such as the `audiomixmatrix` element for example. There are
- also two new functions to set and get properties of this type from bindings:
- - gst_util_set_object_array()
- - gst_util_get_object_array()
-
-- various helper functions have been added to make it easier to set or get
- GstStructure fields containing caps-style array or list fields from language
- bindings (which usually support GValueArray but don't know about the GStreamer
- specific fundamental types):
- - [`gst_structure_get_array()`][get-array]
- - [`gst_structure_set_array()`][set-array]
- - [`gst_structure_get_list()`][get-list]
- - [`gst_structure_set_list()`][set-list]
-
-- a new ['dynamic type' registry factory type][dynamic-type] was added to
- register dynamically loadable GType types. This is useful for automatically
- loading enum/flags types that are used in caps, such as for example the
- `GstVideoMultiviewFlagsSet` type used in multiview video caps.
-
-- there is a new [`GstProxyControlBinding`][proxy-control-binding] for use
- with GstController. This allows proxying the control interface from one
- property on one GstObject to another property (of the same type) in another
- GstObject. So e.g. in parent-child relationship, one may need to call
- `gst_object_sync_values()` on the child and have a binding (set elsewhere)
- on the parent update the value. This is used in `glvideomixer` and `glsinkbin`
- for example, where `sync_values()` on the child pad or element will call
- `sync_values()` on the exposed bin pad or element.
-
- Note that this doesn't solve GObject property forwarding, that must
- be taken care of by the implementation manually or using GBinding.
-
-- `gst_base_parse_drain()` has been made public for subclasses to use.
-
-- `gst_base_sink_set_drop_out_of_segment()' can be used by subclasses to
- prevent GstBaseSink from dropping buffers that fall outside of the segment.
-
-- [`gst_calculate_linear_regression()`][calc-lin-regression] is a new utility
- function to calculate a linear regression.
-
-- [`gst_debug_get_stack_trace`][get-stack-trace] is an easy way to retrieve a
- stack trace, which can be useful in tracer plugins.
-
-- allocators: the dmabuf allocator is now sub-classable, and there is a new
- `GST_CAPS_FEATURE_MEMORY_DMABUF` define.
-
-- video decoder subclasses can use the newly-added function
- `gst_video_decoder_allocate_output_frame_with_params()` to
- pass a `GstBufferPoolAcquireParams` to the buffer pool for
- each buffer allocation.
-
-- the video time code API has gained a dedicated [`GstVideoTimeCodeInterval`][timecode-interval]
- type plus related API, including functions to add intervals to timecodes.
-
-- There is a new `libgstbadallocators-1.0` library in gst-plugins-bad, which
- may go away again in future releases once the `GstPhysMemoryAllocator`
- interface API has been validated by more users and was moved to
- `libgstallocators-1.0` from gst-plugins-base.
-
-[timecode-interval]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstvideo.html#gst-video-time-code-interval-new
-[buffer-iterate-meta-filtered]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstBuffer.html#gst-buffer-iterate-meta-filtered
-[pad-task-get-state]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstPad.html#gst-pad-task-get-state
-[uri-get-fragment-table]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/gstreamer-GstUri.html#gst-uri-get-media-fragment-table
-[print]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/gstreamer-GstInfo.html#gst-print
-[println]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/gstreamer-GstInfo.html#gst-println
-[printerr]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/gstreamer-GstInfo.html#gst-printerr
-[printerrln]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/gstreamer-GstInfo.html#gst-printerrln
-[get-array]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstStructure.html#gst-structure-get-array
-[set-array]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstStructure.html#gst-structure-set-array
-[get-list]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstStructure.html#gst-structure-get-list
-[set-list]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstStructure.html#gst-structure-set-list
-[dynamic-type]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstDynamicTypeFactory.html
-[proxy-control-binding]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer-libs/html/gstreamer-libs-GstProxyControlBinding.html
-[calc-lin-regression]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/gstreamer-GstUtils.html#gst-calculate-linear-regression
-[get-stack-trace]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/gstreamer-GstUtils.html#gst-debug-get-stack-trace
-
-### GstPlayer
-New API has been added to:
+GStreamer OMX
- - get the number of audio/video/subtitle streams:
- - `gst_player_media_info_get_number_of_streams()`
- - `gst_player_media_info_get_number_of_video_streams()`
- - `gst_player_media_info_get_number_of_audio_streams()`
- - `gst_player_media_info_get_number_of_subtitle_streams()`
+- Add support of NV16 format to video encoders input.
- - enable accurate seeking: `gst_player_config_set_seek_accurate()`
- and `gst_player_config_get_seek_accurate()`
+- 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.
- - get a snapshot image of the video in RGBx, BGRx, JPEG, PNG or
- native format: [`gst_player_get_video_snapshot()`][snapshot]
-
- - selecting use of a specific video sink element
- ([`gst_player_video_overlay_video_renderer_new_with_sink()`][renderer-with-vsink])
-
- - If the environment variable `GST_PLAYER_USE_PLAYBIN3` is set, GstPlayer will
- use the still-experimental `playbin3` element and the `GstStreams` API for
- playback.
+- 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.
-[snapshot]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-bad-libs/html/gst-plugins-bad-libs-gstplayer.html#gst-player-get-video-snapshot
-[renderer-with-vsink]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-bad-libs/html/gst-plugins-bad-libs-gstplayer-videooverlayvideorenderer.html#gst-player-video-overlay-video-renderer-new-with-sink
+- Video encoders will now detect frame rate only changes and will
+ inform OMX about it rather than doing a full format reset.
-## Miscellaneous changes
+- 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.
-- video caps for interlaced video may contain an optional `"field-order"` field
- now in the case of `interlaced-mode=interleaved` to signal that the field
- order is always the same throughout the stream. This is useful to signal to
- muxers such as mp4mux. The new field is parsed from/to `GstVideoInfo` of course.
-- video decoder and video encoder base classes try harder to proxy
- interlacing, colorimetry and chroma-site related fields in caps properly.
+GStreamer Editing Services and NLE
-- The buffer stored in the `PROTECTION` events is now left unchanged. This is a
- change of behaviour since 1.8, especially for the mssdemux element which used to
- decode the base64 parsed data wrapped in the protection events emitted by the
- demuxer.
+- this section will be filled in in due course
-- `PROTECTION` events can now be injected into the pipeline from the application;
- source elements deriving from GstBaseSrc will forward those downstream now.
-- The DASH demuxer is now correctly parsing the MSPR-2.0 ContentProtection nodes
- and emits Protection events accordingly. Applications relying on those events
- might need to decode the base64 data stored in the event buffer before using it.
+GStreamer validate
-- The registry can now also be disabled by setting the environment variable
- `GST_REGISTRY_DISABLE=yes`, with similar effect as the `GST_DISABLE_REGISTRY`
- compile time switch.
+- this section will be filled in in due course
-- Seeking performance with gstreamer-vaapi based decoders was improved. It would
- recreate the decoder and surfaces on every seek which can be quite slow.
-- more robust handling of input caps changes in videoaggregator-based elements
- such as `compositor`.
+GStreamer Python Bindings
-- Lots of adaptive streaming-related fixes across the board (DASH, MSS, HLS). Also:
+- add binding for gst_pad_set_caps()
- - `mssdemux`, the Microsoft Smooth Streaming demuxer, has seen various
- fixes for live streams, duration reporting and seeking.
+- pygobject dependency requirement was bumped to >= 3.8
- - The DASH manifest parser now extracts MS PlayReady ContentProtection objects
- from manifests and sends them downstream as `PROTECTION` events. It also
- supports multiple Period elements in external xml now.
+- new audiotestsrc, audioplot, and mixer plugin examples, and a
+ dynamic pipeline example
-- gst-libav was updated to ffmpeg 3.3 but should still work with any 3.x
- version.
-- GstEncodingProfile has been generally enhanced so it can, for
- example, be used to get possible profiles for a given file
- extension. It is now possible to define profiles based on element
- factory names or using a path to a `.gep` file containing a
- serialized profile.
+GStreamer C# Bindings
-- `audioconvert` can now do endianness conversion in-place. All other
- conversions still require a copy, but e.g. sign conversion and a few others
- could also be implemented in-place now.
+- bindings for the GstWebRTC library
-- The new, experimental `playbin3` and `urisourcebin` elements got many
- bugfixes and improvements and should generally be closer to a full
- replacement of the old elements.
-- `interleave` now supports > 64 channels.
+GStreamer Rust Bindings
-### OpenGL integration
+The GStreamer Rust bindings are now officially part of the GStreamer
+project and are also maintained in the GStreamer GitLab.
-- As usual the GStreamer OpenGL integration library has seen numerous
- fixes and performance improvements all over the place, and is hopefully
- ready now to become API stable and be moved to gst-plugins-base during the
- 1.14 release cycle.
+The releases will generally not be synchronized with the releases of
+other GStreamer parts due to dependencies on other projects.
-- The GStreamer OpenGL integration layer has also gained support for the
- Vivante EGL FB windowing system, which improves performance on platforms
- such as Freescale iMX.6 for those who are stuck with the proprietary driver.
- The `qmlglsink` element also supports this now if Qt is used with eglfs or
- wayland backend, and it works in conjunction with [gstreamer-imx][gstreamer-imx]
- of course.
+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.:
-- various `qmlglsrc` improvements
+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.
-[gstreamer-imx]: https://github.com/Freescale/gstreamer-imx
+The current stable release is 0.12.2 and the next release series will be
+0.13, probably around March 2019.
-## Tracing framework and debugging improvements
+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.
-- New tracing hooks have been added to track GstMiniObject and GstObject
- ref/unref operations.
+Also included is support for creating subclasses of the following types
+and writing GStreamer plugins:
-- The memory leaks tracer can optionally use this to retrieve stack traces if
- enabled with e.g. `GST_TRACERS=leaks(filters="GstEvent,GstMessage",stack-traces-flags=full)`
+- 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
-- The `GST_DEBUG_FILE` environment variable, which can be used to write the
- debug log output to a file instead of printing it to stderr, can now contain
- a name pattern, which is useful for automated testing and continuous
- integration systems. The following format specifiers are supported:
+Changes to 0.12.X since 0.12.0
- - `%p`: will be replaced with the PID
- - `%r`: will be replaced with a random number, which is useful for instance
- when running two processes with the same PID but in different containers.
+Fixed
-## Tools
+- PTP clock constructor actually creates a PTP instead of NTP clock
-- `gst-inspect-1.0` can now list elements by type with the new `--types`
- command-line option, e.g. `gst-inspect-1.0 --types=Audio/Encoder` will
- show a list of audio encoders.
+Added
-- `gst-launch-1.0` and `gst_parse_launch()` have gained a new operator (`:`)
- that allows linking all pads between two elements. This is useful in cases
- where the exact number of pads or type of pads is not known beforehand, such
- as in the `uridecodebin : encodebin` scenario, for example. In this case,
- multiple links will be created if the encodebin has multiple profiles
- compatible with the output of uridecodebin.
+- Bindings for GStreamer Editing Services
+- Bindings for GStreamer Check testing library
+- Bindings for the encoding profile API (encodebin)
-- `gst-device-monitor-1.0` now shows a `gst-launch-1.0` snippet for each
- device that shows how to make use of it in a `gst-launch-1.0` pipeline string.
+- 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()
-## GStreamer RTSP server
+Changed
-- The RTSP server now also supports Digest authentication in addition to Basic
- authentication.
+- 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
-- The `GstRTSPClient` class has gained a `pre-*-request` signal and virtual
- method for each client request type, emitted in the beginning of each rtsp
- request. These signals or virtual methods let the application validate the
- requests, configure the media/stream in a certain way and also generate error
- status codes in case of an error or a bad request.
+Changes in git master for 0.13
-## GStreamer VAAPI
+Fixed
-- GstVaapiDisplay now inherits from GstObject, thus the VA display logging
- messages are better and tracing the context sharing is more readable.
+- gst::tag::Album is the album tag now instead of artist sortname
-- When uploading raw images into a VA surfaces now VADeriveImages are tried
- fist, improving the upload performance, if it is possible.
+Added
-- The decoders and the post-processor now can push dmabuf-based buffers to
- downstream under certain conditions. For example:
+- 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.
- `GST_GL_PLATFORM=egl gst-play-1.0 video-sample.mkv --videosink=glimagesink`
+- Bindings for CapsFeatures and Meta
+- Bindings for
+ ParentBufferMeta,VideoMetaandVideoOverlayCompositionMeta`
+- Bindings for VideoOverlayComposition and VideoOverlayRectangle
+- Bindings for VideoTimeCode
-- Refactored the wrapping of VA surface into gstreamer memory, adding lock
- when mapping and unmapping, and many other fixes.
+- 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
-- Now `vaapidecodebin` loads `vaapipostproc` dynamically. It is possible to
- avoid it usage with the environment variable `GST_VAAPI_DISABLE_VPP=1`.
+- 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
-- Regarding encoders: they have primary rank again, since they can discover,
- in run-time, the color formats they can use for upstream raw buffers and
- caps renegotiation is now possible. Also the encoders push encoding info
- downstream via tags.
+- serde serialization of Value can also handle Buffer now
-- About specific encoders: added constant bit-rate encoding mode for VP8 and
- H265 encoder handles P010_10LE color format.
+- 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
-- Regarding decoders, flush operation has been improved, now the internal VA
- encoder is not recreated at each flush. Also there are several improvements
- in the handling of H264 and H265 streams.
+Changed
-- VAAPI plugins try to create their on GstGL context (when available) if they
- cannot find it in the pipeline, to figure out what type of VA Display they
- should create.
+- Rust 1.31 is the minimum supported Rust version now
+- Update to latest gir code generator and glib bindings
-- Regarding `vaapisink` for X11, if the backend reports that it is unable to
- render correctly the current color format, an internal VA post-processor, is
- instantiated (if available) and converts the color format.
+- 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.
-## GStreamer Editing Services and NLE
+- 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.
-- Enhanced auto transition behaviour
+- BinExt extension trait was renamed to GstBinExt to prevent conflicts
+ with gtk::Bin if both are imported
-- Fix some races in `nlecomposition`
+- Buffer::from_slice() can’t possible return None
-- Allow building with msvc
+- Various clippy warnings
-- Added a UNIX manpage for `ges-launch`
-- API changes:
- - Added ges_deinit (allowing the leak tracer to work properly)
- - Added ges_layer_get_clips_in_interval
- - Finally hide internal symbols that should never have been exposed
+GStreamer Rust Plugins
-## GStreamer validate
+Like the GStreamer Rust bindings, the Rust plugins are now officially
+part of the GStreamer project and are also maintained in the GStreamer
+GitLab.
-- Port `gst-validate-launcher` to python 3
+In the 0.3.x versions this contained infrastructure for writing
+GStreamer plugins in Rust, and a set of plugins.
-- `gst-validate-launcher` now checks if blacklisted bugs have been fixed on
- bugzilla and errors out if it is the case
+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.
-- Allow building with msvc
+Elements included are:
-- Add ability for the launcher to run GStreamer unit tests
+- Tutorials plugin: identity, rgb2gray and sinesrc with extensive
+ comments
-- Added a way to activate the leaks tracer on our tests and fix leaks
+- rsaudioecho, a port of the audiofx element
-- Make the http server multithreaded
+- rsfilesrc, rsfilesink
-- New testsuite for running various test scenarios on the DASH-IF test vectors
+- rsflvdemux, a FLV demuxer. Not feature-equivalent with flvdemux yet
-## Build and Dependencies
+- 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.
-- Meson build files are now disted in tarballs, for jhbuild and so distro
- packagers can start using it. Note that the Meson-based build system is not
- 100% feature-equivalent with the autotools-based one yet.
+- rshttpsrc, a HTTP source around the hyper/reqwest Rust libraries.
+ Not feature-equivalent with souphttpsrc yet.
-- Some plugin filenames have been changed to match the plugin names: for example
- the file name of the `encoding` plugin in gst-plugins-base containing the
- `encodebin` element was `libgstencodebin.so` and has been changed to
- `libgstencodebin.so`. This affects only a handful of plugins across modules.
+- togglerecord, an element that allows to start/stop recording at any
+ time and keeps all audio/video streams in sync.
- **Developers who install GStreamer from source and just do `make install`**
- **after updating the source code, without doing `make uninstall` first, will**
- **have to manually remove the old installed plugin files from the installation**
- **prefix, or they will get 'Cannot register existing type' critical warnings.**
+- mccparse and mccenc, parsers and encoders for the MCC closed caption
+ file format.
-- Most of the docbook-based documentation (FAQ, Application Development Manual,
- Plugin Writer's Guide, design documents) has been converted to markdown and
- moved into a new gst-docs module. The gtk-doc library API references and
- the plugins documentation are still built as part of the source modules though.
+Changes to 0.3.X since 0.3.0
-- GStreamer core now optionally uses libunwind and libdw to generate backtraces.
- This is useful for tracer plugins used during debugging and development.
+- 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
-- There is a new `libgstbadallocators-1.0` library in gst-plugins-bad (which
- may go away again in future releases once the `GstPhysMemoryAllocator`
- interface API has been validated by more users).
+Changes in git master for 0.4
-- `gst-omx` and `gstreamer-vaapi` modules can now also be built using the
- Meson build system.
+- 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
-- The `qtkitvideosrc` element for macOS was removed. The API is deprecated
- since 10.9 and it wasn't shipped in the binaries since a few releases.
+- Register plugins statically in tests instead of relying on the
+ plugin loader to find the shared library in a specific place
-## Platform-specific improvements
+- Update to the latest API changes in the GLib and GStreamer bindings
+- Update to the latest versions of all crates
-### Android
-- androidmedia: add support for VP9 video decoding/encoding and Opus audio
- decoding (where supported)
+Build and Dependencies
-### OS/X and iOS
+- 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
-- `avfvideosrc`, which represents an iPhone camera or, on a Mac, a screencapture
- session, so far allowed you to select an input device by device index only.
- New API adds the ability to select the position (front or back facing) and
- device-type (wide angle, telephoto, etc.). Furthermore, you can now also
- specify the orientation (portrait, landscape, etc.) of the videostream.
+- 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.
-### Windows
+- 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.
-- `dx9screencapsrc` can now optionally also capture the cursor.
+- 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.
-## Contributors
+- The (optional) OpenCV requirement has been bumped to >= 3.0.0 and
+ the plugin can also be built against OpenCV 4.x now.
-Aleix Conchillo Flaque, Alejandro G. Castro, Aleksandr Slobodeniuk, Alexandru
-Băluț, Alex Ashley, Andre McCurdy, Andrew, Anton Eliasson, Antonio Ospite,
-Arnaud Vrac, Arun Raghavan, Aurélien Zanelli, Axel Menzel, Benjamin Otte,
-Branko Subasic, Brendan Shanks, Carl Karsten, Carlos Rafael Giani, ChangBok
-Chae, Chris Bass, Christian Schaller, christophecvr, Claudio Saavedra,
-Corentin Noël, Dag Gullberg, Daniel Garbanzo, Daniel Shahaf, David Evans,
-David Schleef, David Warman, Dominique Leuenberger, Dongil Park, Douglas
-Bagnall, Edgard Lima, Edward Hervey, Emeric Grange, Enrico Jorns, Enrique
-Ocaña González, Evan Nemerson, Fabian Orccon, Fabien Dessenne, Fabrice Bellet,
-Florent Thiéry, Florian Zwoch, Francisco Velazquez, Frédéric Dalleau, Garima
-Gaur, Gaurav Gupta, George Kiagiadakis, Georg Lippitsch, Göran Jönsson, Graham
-Leggett, Guillaume Desmottes, Gurkirpal Singh, Haihua Hu, Hanno Boeck, Havard
-Graff, Heekyoung Seo, hoonhee.lee, Hyunjun Ko, Imre Eörs, Iñaki García
-Etxebarria, Jagadish, Jagyum Koo, Jan Alexander Steffens (heftig), Jan
-Schmidt, Jean-Christophe Trotin, Jochen Henneberg, Jonas Holmberg, Joris
-Valette, Josep Torra, Juan Pablo Ugarte, Julien Isorce, Jürgen Sachs, Koop
-Mast, Kseniia Vasilchuk, Lars Wendler, leigh123linux@googlemail.com, Luis de
-Bethencourt, Lyon Wang, Marcin Kolny, Marinus Schraal, Mark Nauwelaerts,
-Mathieu Duponchelle, Matthew Waters, Matt Staples, Michael Dutka, Michael
-Olbrich, Michael Smith, Michael Tretter, Miguel París Díaz, namanyadav12, Neha
-Arora, Nick Kallen, Nicola Murino, Nicolas Dechesne, Nicolas Dufresne, Nicolas
-Huet, Nirbheek Chauhan, Ole André Vadla Ravnås, Olivier Crête, Patricia
-Muscalu, Peter Korsgaard, Peter Seiderer, Petr Kulhavy, Philippe Normand,
-Philippe Renon, Philipp Zabel, Rahul Bedarkar, Reynaldo H. Verdejo Pinochet,
-Ricardo Ribalda Delgado, Rico Tzschichholz, Руслан Ижбулатов, Samuel Maroy,
-Santiago Carot-Nemesio, Scott D Phillips, Sean DuBois, Sebastian Dröge, Sergey
-Borovkov, Seungha Yang, shakin chou, Song Bing, Søren Juul, Sreerenj
-Balachandran, Stefan Kost, Stefan Sauer, Stepan Salenikovich, Stian Selnes,
-Stuart Weaver, suhas2go, Thiago Santos, Thibault Saunier, Thomas Bluemel,
-Thomas Petazzoni, Tim-Philipp Müller, Ting-Wei Lan, Tobias Mueller, Todor
-Tomov, Tomasz Zajac, Ulf Olsson, Ursula Maplehurst, Víctor Manuel Jáquez Leal,
-Victor Toso, Vincent Penquerc'h, Vineeth TM, Vinod Kesti, Vitor Massaru Iha,
-Vivia Nikolaidou, WeiChungChang, William Manley, Wim Taymans, Wojciech
-Przybyl, Wonchul Lee, Xavier Claessens, Yasushi SHOJI
-
-... and many others who have contributed bug reports, translations, sent
+- New sctp plugin based on usrsctp (for WebRTC data channels)
+
+Cerbero
+
+Cerbero is a meta build system used to build GStreamer plus dependencies
+on platforms where dependencies are not readily available, such as
+Windows, Android, iOS and macOS.
+
+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
+
+
+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.
+
+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.
+
+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
+
+- 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
+
+
+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,
+
+… and many others who have contributed bug reports, translations, sent
suggestions or helped testing.
-## Bugs fixed in 1.12
-More than [635 bugs][bugs-fixed-in-1.12] have been fixed during
-the development of 1.12.
+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.
This list does not include issues that have been cherry-picked into the
-stable 1.10 branch and fixed there as well, all fixes that ended up in the
-1.10 branch are also included in 1.12.
+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.
+
-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
-[bugs-fixed-in-1.12]: https://bugzilla.gnome.org/buglist.cgi?bug_status=RESOLVED&bug_status=VERIFIED&classification=Platform&limit=0&list_id=213265&order=bug_id&product=GStreamer&query_format=advanced&resolution=FIXED&target_milestone=1.10.1&target_milestone=1.10.2&target_milestone=1.10.3&target_milestone=1.10.4&target_milestone=1.11.1&target_milestone=1.11.2&target_milestone=1.11.3&target_milestone=1.11.4&target_milestone=1.11.90&target_milestone=1.11.91&target_milestone=1.12.0
+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 1.16.x bug-fix releases will be made from
+the git 1.16 branch, which is a stable branch.
-## Stable 1.12 branch
+1.16.0
-After the 1.12.0 release there will be several 1.12.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 1.12.x bug-fix releases will be made from the git 1.12 branch, which
-is a stable branch.
+1.16.0 is scheduled to be released in March 2019.
-### 1.12.0
-1.12.0 was released on 4th May 2017.
+Known Issues
-## Known Issues
+- possibly breaking/incompatible changes to properties of wrapped
+ FFmpeg decoders and encoders (see above).
-- The `webrtcdsp` element is currently not shipped as part of the Windows
- binary packages due to a [build system issue][bug-770264].
+- 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.
-[bug-770264]: https://bugzilla.gnome.org/show_bug.cgi?id=770264
-## Schedule for 1.14
+Schedule for 1.18
-Our next major feature release will be 1.14, and 1.11 will be the unstable
-development version leading up to the stable 1.12 release. The development
-of 1.13/1.14 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 1.14 development cycle is yet to be confirmed, but it is
-expected that feature freeze will be around September 2017
-followed by several 1.13 pre-releases and the new 1.14 stable release
-in October.
+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.
-1.14 will be backwards-compatible to the stable 1.12, 1.10, 1.8, 1.6, 1.4,
-1.2 and 1.0 release series.
+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.
-- - -
+------------------------------------------------------------------------
-*These release notes have been prepared by Sebastian Dröge, Tim-Philipp Müller
-and Víctor Manuel Jáquez Leal.*
+_These release notes have been prepared by Tim-Philipp Müller with_
+_contributions from Sebastian Dröge and Guillaume Desmottes._
-*License: [CC BY-SA 4.0](http://creativecommons.org/licenses/by-sa/4.0/)*
+_License: CC BY-SA 4.0_
projects / platform / upstream / gstreamer.git / blobdiff