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@@ -7,13 +7,13 @@ GStreamer 1.14.0 has not been released yet. It is scheduled for release
in early March 2018.
There are unstable pre-releases available for testing and development
-purposes. The latest pre-release is version 1.13.90 (rc1) and was
-released on 03 March 2018.
+purposes. The latest pre-release is version 1.13.91 (rc2) and was
+released on 12 March 2018.
See https://gstreamer.freedesktop.org/releases/1.14/ for the latest
version of this document.
-_Last updated: Saturday 03 March 2018, 16:30 UTC (log)_
+_Last updated: Monday 12 March 2018, 18:00 UTC (log)_
Introduction
@@ -28,103 +28,957 @@ other improvements.
Highlights
-- this section will be completed shortly
+- WebRTC support: real-time audio/video streaming to and from web
+ browsers
+
+- Experimental support for the next-gen royalty-free AV1 video codec
+
+- Video4Linux: encoding support, stable element names and faster
+ device probing
+
+- Support for the Secure Reliable Transport (SRT) video streaming
+ protocol
+
+- RTP Forward Error Correction (FEC) support (ULPFEC)
+
+- RTSP 2.0 support in rtspsrc and gst-rtsp-server
+
+- ONVIF audio backchannel support in gst-rtsp-server and rtspsrc
+
+- playbin3 gapless playback and pre-buffering support
+
+- tee, our stream splitter/duplication element, now does allocation
+ query aggregation which is important for efficient data handling and
+ zero-copy
+
+- QuickTime muxer has a new prefill recording mode that allows file
+ import in Adobe Premiere and FinalCut Pro while the file is still
+ being written.
+
+- rtpjitterbuffer fast-start mode and timestamp offset adjustment
+ smoothing
+
+- souphttpsrc connection sharing, which allows for connection reuse,
+ cookie sharing, etc.
+
+- nvdec: new plugin for hardware-accelerated video decoding using the
+ NVIDIA NVDEC API
+
+- Adaptive DASH trick play support
+
+- ipcpipeline: new plugin that allows splitting a pipeline across
+ multiple processes
+
+- Major gobject-introspection annotation improvements for large parts
+ of the library API
Major new features and changes
-Noteworthy new API
+WebRTC support
+
+There is now basic support for WebRTC in GStreamer in form of a new
+webrtcbin element and a webrtc support library. This allows you to build
+applications that set up connections with and stream to and from other
+WebRTC peers, whilst leveraging all of the usual GStreamer features such
+as hardware-accelerated encoding and decoding, OpenGL integration,
+zero-copy and embedded platform support. And it's easy to build and
+integrate into your application too!
+
+WebRTC enables real-time communication of audio, video and data with web
+browsers and native apps, and it is supported or about to be support by
+recent versions of all major browsers and operating systems.
-- this section will be filled in shortly
+GStreamer's new WebRTC implementation uses libnice for Interactive
+Connectivity Establishment (ICE) to figure out the best way to
+communicate with other peers, punch holes into firewalls, and traverse
+NATs.
+
+The implementation is not complete, but all the basics are there, and
+the code sticks fairly close to the PeerConnection API. Where
+functionality is missing it should be fairly obvious where it needs to
+go.
+
+For more details, background and example code, check out Nirbheek's blog
+post _GStreamer has grown a WebRTC implementation_, as well as Matthew's
+_GStreamer WebRTC_ talk from last year's GStreamer Conference in Prague.
New Elements
-- this section will be filled in shortly
+- webrtcbin handles the transport aspects of webrtc connections (see
+ WebRTC section above for more details)
+
+- New srtsink and srtsrc elements for the Secure Reliable Transport
+ (SRT) video streaming protocol, which aims to be easy to use whilst
+ striking a new balance between reliability and latency for low
+ latency video streaming use cases. More details about SRT and the
+ implementation in GStreamer in Olivier's blog post _SRT in
+ GStreamer_.
+
+- av1enc and av1dec elements providing experimental support for the
+ next-generation royalty free video AV1 codec, alongside Matroska
+ support for it.
+
+- hlssink2 is a rewrite of the existing hlssink element, but unlike
+ its predecessor hlssink2 takes elementary streams as input and
+ handles the muxing to MPEG-TS internally. It also leverages
+ splitmuxsink internally to do the splitting. This allows more
+ control over the chunk splitting and sizing process and relies less
+ on the co-operation of an upstream muxer. Different to the old
+ hlssink it also works with pre-encoded streams and does not require
+ close interaction with an upstream encoder element.
+
+- audiolatency is a new element for measuring audio latency end-to-end
+ and is useful to measure roundtrip latency including both the
+ GStreamer-internal latency as well as latency added by external
+ components or circuits.
+
+- 'fakevideosink is basically a null sink for video data and very
+ similar to fakesink, only that it will answer allocation queries and
+ will advertise support for various video-specific things such
+ GstVideoMeta, GstVideoCropMeta and GstVideoOverlayCompositionMeta
+ like a normal video sink would. This is useful for throughput
+ testing and testing the zero-copy path when creating a new pipeline.
+
+- ipcpipeline: new plugin that allows the splitting of a pipeline into
+ multiple processes. Usually a GStreamer pipeline runs in a single
+ process and parallelism is achieved by distributing workloads using
+ multiple threads. This means that all elements in the pipeline have
+ access to all the other elements' memory space however, including
+ that of any libraries used. For security reasons one might therefore
+ want to put sensitive parts of a pipeline such as DRM and decryption
+ handling into a separate process to isolate it from the rest of the
+ pipeline. This can now be achieved with the new ipcpipeline plugin.
+ Check out George's blog post _ipcpipeline: Splitting a GStreamer
+ pipeline into multiple processes_ or his lightning talk from last
+ year's GStreamer Conference in Prague for all the gory details.
+
+ 
+- proxysink and proxysrc are new elements to pass data from one
+ pipeline to another within the same process, very similar to the
+ existing inter elements, but not limited to raw audio and video
+ data. These new proxy elements are very special in how they work
+ under the hood, which makes them extremely powerful, but also
+ dangerous if not used with care. The reason for this is that it's
+ not just data that's passed from sink to src, but these elements
+ basically establish a two-way wormhole that passes through queries
+ and events in both directions, which means caps negotiation and
+ allocation query driven zero-copy can work through this wormhole.
+ There are scheduling considerations as well: proxysink forwards
+ everything into the proxysrc pipeline directly from the proxysink
+ streaming thread. There is a queue element inside proxysrc to
+ decouple the source thread from the sink thread, but that queue is
+ not unlimited, so it is entirely possible that the proxysink
+ pipeline thread gets stuck in the proxysrc pipeline, e.g. when that
+ pipeline is paused or stops consuming data for some other reason.
+ This means that one should always shut down down the proxysrc
+ pipeline before shutting down the proxysink pipeline, for example.
+ Or at least take care when shutting down pipelines. Usually this is
+ not a problem though, especially not in live pipelines. For more
+ information see Nirbheek's blog post _Decoupling GStreamer
+ Pipelines_, and also check out out the new ipcpipeline plugin for
+ sending data from one process to another process (see above).
+
+- lcms is a new LCMS-based ICC color profile correction element
+
+- openmptdec is a new OpenMPT-based decoder for module music formats,
+ such as S3M, MOD, XM, IT. It is built on top of a new
+ GstNonstreamAudioDecoder base class which aims to unify handling of
+ files which do not operate a streaming model. The wildmidi plugin
+ has also been revived and is also implemented on top of this new
+ base class.
+
+- The curl plugin has gained a new curlhttpsrc element, which is
+ useful for testing HTTP protocol version 2.0 amongst other things.
-New element features and additions
+Noteworthy new API
-- this section will be filled in shortly
+- GstPromise provides future/promise-like functionality. This is used
+ in the GStreamer WebRTC implementation.
+
+ 
+- GstReferenceTimestampMeta is a new meta that allows you to attach
+ additional reference timestamps to a buffer. These timestamps don't
+ have to relate to the pipeline clock in any way. Examples of this
+ could be an NTP timestamp when the media was captured, a frame
+ counter on the capture side or the (local) UNIX timestamp when the
+ media was captured. The decklink elements make use of this.
+
+ 
+- GstVideoRegionOfInterestMeta: it's now possible to attach generic
+ free-form element-specific parameters to a region of interest meta,
+ for example to tell a downstream encoder to use certain codec
+ parameters for a certain region.
+
+ 
+- gst_bus_get_pollfd can be used to obtain a file descriptor for the
+ bus that can be poll()-ed on for new messages. This is useful for
+ integration with non-GLib event loops.
+
+ 
+- gst_get_main_executable_path() can be used by wrapper plugins that
+ need to find things in the directory where the application
+ executable is located. In the same vein,
+ GST_PLUGIN_DEPENDENCY_FLAG_PATHS_ARE_RELATIVE_TO_EXE can be used to
+ signal that plugin dependency paths are relative to the main
+ executable.
+
+- pad templates can be told about the GType of the pad subclass of the
+ pad via newly-added GstPadTemplate API API or the
+ gst_element_class_add_static_pad_template_with_gtype() convenience
+ function. gst-inspect-1.0 will use this information to print pad
+ properties.
+
+ 
+- new convenience functions to iterate over element pads without using
+ the GstIterator API: gst_element_foreach_pad(),
+ gst_element_foreach_src_pad(), and gst_element_foreach_sink_pad().
+
+ 
+- GstBaseSrc and appsrc have gained support for buffer lists:
+ GstBaseSrc subclasses can use gst_base_src_submit_buffer_list(), and
+ applications can use gst_app_src_push_buffer_list() to push a buffer
+ list into appsrc.
+
+ 
+- The GstHarness unit test harness has a couple of new convenience
+ functions to retrieve all pending data in the harness in form of a
+ single chunk of memory.
+
+ 
+- GstAudioStreamAlign is a new helper object for audio elements that
+ handles discontinuity detection and sample alignment. It will align
+ samples after the previous buffer's samples, but keep track of the
+ divergence between buffer timestamps and sample position (jitter).
+ If it exceeds a configurable threshold the alignment will be reset.
+ This simply factors out code that was duplicated in a number of
+ elements into a common helper API.
+
+ 
+- The GstVideoEncoder base class implements Quality of Service (QoS)
+ now. This is disabled by default and must be opted in by setting the
+ "qos" property, which will make the base class gather statistics
+ about the real-time performance of the pipeline from downstream
+ elements (usually sinks that sync the pipeline clock). Subclasses
+ can then make use of this by checking whether input frames are late
+ already using gst_video_encoder_get_max_encode_time() If late, they
+ can just drop them and skip encoding in the hope that the pipeline
+ will catch up.
+
+ 
+- The GstVideoOverlay interface gained a few helper functions for
+ installing and handling a "render-rectangle" property on elements
+ that implement this interface, so that this functionality can also
+ be used from the command line for testing and debugging purposes.
+ The property wasn't added to the interface itself as that would
+ require all implementors to provide it which would not be
+ backwards-compatible.
+
+ 
+- A new base class, GstNonstreamAudioDecoder for non-stream audio
+ decoders was added to gst-plugins-bad. This base-class is meant to
+ be used for audio decoders that require the whole stream to be
+ loaded first before decoding can start. Examples of this are module
+ formats (MOD/S3M/XM/IT/etc), C64 SID tunes, video console music
+ files (GYM/VGM/etc), MIDI files and others. The new openmptdec
+ element is based on this.
+
+ 
+- Full list of API new in 1.14:
+- GStreamer core API new in 1.14
+- GStreamer base library API new in 1.14
+- gst-plugins-base libraries API new in 1.14
+- gst-plugins-bad: no list, mostly GstWebRTC library and new
+ non-stream audio decoder base class.
+
+New RTP features and improvements
+
+- rtpulpfecenc and rtpulpfecdec are new elements that implement
+ Generic Forward Error Correction (FEC) using Uneven Level Protection
+ (ULP) as described in RFC 5109. This can be used to protect against
+ certain types of (non-bursty) packet loss, and important packets
+ such as those containing codec configuration data or key frames can
+ be protected with higher redundancy. Equally, packets that are not
+ particularly important can be given low priority or not be protected
+ at all. If packets are lost, the receiver can then hopefully restore
+ the lost packet(s) from the surrounding packets which were received.
+ This is an alternative to, or rather complementary to, dealing with
+ packet loss using _retransmission (rtx)_. GStreamer has had
+ retransmission support for a long time, but Forward Error Correction
+ allows for different trade-offs: The advantage of Forward Error
+ Correction is that it doesn't add latency, whereas retransmission
+ requires at least one more roundtrip to request and hopefully
+ receive lost packets; Forward Error Correction increases the
+ required bandwidth however, even in situations where there is no
+ packet loss at all, so one will typically want to fine-tune the
+ overhead and mechanisms used based on the characteristics of the
+ link at the time.
+
+- New _Redundant Audio Data (RED)_ encoders and decoders for RTP as
+ per RFC 2198 are also provided (rtpredenc and rtpreddec), mostly for
+ chrome webrtc compatibility, as chrome will wrap ULPFEC-protected
+ streams in RED packets, and such streams need to be wrapped and
+ unwrapped in order to use ULPFEC with chrome.
+
+ 
+- a few new buffer flags for FEC support:
+ GST_BUFFER_FLAG_NON_DROPPABLE can be used to mark important buffers,
+ e.g. to flag RTP packets carrying keyframes or codec setup data for
+ RTP Forward Error Correction purposes, or to prevent still video
+ frames from being dropped by elements due to QoS. There already is a
+ GST_BUFFER_FLAG_DROPPABLE. GST_RTP_BUFFER_FLAG_REDUNDANT is used to
+ signal internally that a packet represents a redundant RTP packet
+ and used in rtpstorage to hold back the packet and use it only for
+ recovery from packet loss. Further work is still needed in
+ payloaders to make use of these.
+
+- rtpbin now has an option for increasing timestamp offsets gradually:
+ Instant large changes to the internal ts_offset may cause timestamps
+ to move backwards and also cause visible glitches in media playback.
+ The new "max-ts-offset-adjustment" and "max-ts-offset" properties
+ let the application control the rate to apply changes to ts_offset.
+ There have also been some EOS/BYE handling improvements in rtpbin.
+
+- rtpjitterbuffer has a new fast start mode: in many scenarios the
+ jitter buffer will have to wait for the full configured latency
+ before it can start outputting packets. The reason for that is that
+ it often can't know what the sequence number of the first expected
+ RTP packet is, so it can't know whether a packet earlier than the
+ earliest packet received will still arrive in future. This behaviour
+ can now be bypassed by setting the "faststart-min-packets" property
+ to the number of consecutive packets needed to start, and the jitter
+ buffer will start output packets as soon as it has N consecutive
+ packets queued internally. This is particularly useful to get a
+ first video frame decoded and rendered as quickly as possible.
+
+- rtpL8pay and rtpL8depay provide RTP payloading and depayloading for
+ 8-bit raw audio
+
+New element features
+
+- playbin3 has gained support or gapless playback via the
+ "about-to-finish" signal where users can set the uri for the next
+ item to play. For non-live streams this will be emitted as soon as
+ the first uri has finished downloading, so with sufficiently large
+ buffers it is now possible to pre-buffer the next item well ahead of
+ time (unlike playbin where there would not be a lot of time between
+ "about-to-finish" emission and the end of the stream). If the stream
+ format of the next stream is the same as that of the previous
+ stream, the data will be concatenated via the concat element.
+ Whether this will result in true gaplessness depends on the
+ container format and codecs used, there might still be codec-related
+ gaps between streams with some codecs.
+
+- tee now does allocation query aggregation, which is important for
+ zero-copy and efficient data handling, especially for video. Those
+ who want to drop allocation queries on purpose can use the identity
+ element's new "drop-allocation" property for that instead.
+
+- audioconvert now has a "mix-matrix" property, which obsoletes the
+ audiomixmatrix element. There's also mix matrix support in the audio
+ conversion and channel mixing API.
+
+- x264enc: new "insert-vui" property to disable VUI (Video Usability
+ Information) parameter insertion into the stream, which allows
+ creation of streams that are compatible with certain legacy hardware
+ decoders that will refuse to decode in certain combinations of
+ resolution and VUI parameters; the max. allowed number of B-frames
+ was also increased from 4 to 16.
+
+- dvdlpcmdec: has gained support for Blu-Ray audio LPCM.
+
+- appsrc has gained support for buffer lists (see above) and also seen
+ some other performance improvements.
+
+- flvmux has been ported to the GstAggregator base class which means
+ it can work in defined-latency mode with live input sources and
+ continue streaming if one of the inputs stops producing data.
+
+- jpegenc has gained a "snapshot" property just like pngenc to make it
+ easier to just output a single encoded frame.
+
+- jpegdec will now handle interlaced MJPEG streams properly and also
+ handle frames without an End of Image marker better.
+
+- v4l2: There are now video encoders for VP8, VP9, MPEG4, and H263.
+ The v4l2 video decoder handles dynamic resolution changes, and the
+ video4linux device provider now does much faster device probing. The
+ plugin also no longer uses the libv4l2 library by default, as it has
+ prevented a lot of interesting use cases like CREATE_BUFS, DMABuf,
+ usage of TRY_FMT. As the libv4l2 library is totally inactive and not
+ really maintained, we decided to disable it. This might affect a
+ small number of cheap/old webcams with custom vendor formats for
+ which we do not provide conversion in GStreamer. It is possible to
+ re-enable support for libv4l2 at run-time however, by setting the
+ environment variable GST_V4L2_USE_LIBV4L2=1.
+
+- rtspsrc now has support for RTSP protocol version 2.0 as well as
+ ONVIF audio backchannels (see below for more details). It also
+ sports a new ["accept-certificate"] signal for "manually" checking a
+ TLS certificate for validity. It now also prints RTSP/SDP messages
+ to the gstreamer debug log instead of stdout.
+
+- shout2send now uses non-blocking I/O and has a configurable network
+ operations timeout.
+
+- splitmuxsink has gained a "split-now" action signal and new
+ "alignment-threshold" and "use-robust-muxing" properties. If robust
+ muxing is enabled, it will check and set the muxer's reserved space
+ properties if present. This is primarily for use with mp4mux's
+ robust muxing mode.
+
+- qtmux has a new _prefill recording mode_ which sets up a moov header
+ with the correct sample positions beforehand, which then allows
+ software like Adobe Premiere and FinalCut Pro to import the files
+ while they are still being written to. This only works with constant
+ framerate I-frame only streams, and for now only support for ProRes
+ video and raw audio is implemented but adding new codecs is just a
+ matter of defining appropriate maximum frame sizes.
+
+- qtmux also supports writing of svmi atoms with stereoscopic video
+ information now. Trak timescales can be configured on a per-stream
+ basis using the "trak-timescale" property on the sink pads. Various
+ new formats can be muxed: MPEG layer 1 and 2, AC3 and Opus, as well
+ as PNG and VP9.
+
+- souphttpsrc now does connection sharing by default, shares its
+ SoupSession with other elements in the same pipeline via a
+ GstContext if possible (session-wide settings are all the defaults).
+ This allows for connection reuse, cookie sharing, etc. Applications
+ can also force a context to use. In other news, HTTP headers
+ received from the server are posted as element messages on the bus
+ now for easier diagnostics, and it's also possible now to use other
+ types of proxy servers such as SOCKS4 or SOCKS5 proxies, support for
+ which is implemented directly in gio. Before only HTTP proxies were
+ allowed.
+
+- qtmux, mp4mux and matroskamux will now refuse caps changes of input
+ streams at runtime. This isn't really supported with these
+ containers (or would have to be implemented differently with a
+ considerable effort) and doesn't produce valid and spec-compliant
+ files that will play everywhere. So if you can't guarantee that the
+ input caps won't change, use a container format that does support on
+ the fly caps changes for a stream such as MPEG-TS or use
+ splitmuxsink which can start a new file when the caps change. What
+ would happen before is that e.g. rtph264depay or rtph265depay would
+ simply send new SPS/PPS inband even for AVC format, which would then
+ get muxed into the container as if nothing changed. Some decoders
+ will handle this just fine, but that's often more luck than by
+ design. In any case, it's not right, so we disallow it now.
+
+- matroskamux had Table of Content (TOC) support now (chapters etc.)
+ and matroskademux TOC support has been improved. matroskademux has
+ also seen seeking improvements searching for the right cluster and
+ position.
+
+- videocrop now uses GstVideoCropMeta if downstream supports it, which
+ means cropping can be handled more efficiently without any copying.
+
+- compositor now has support for _crossfade blending_, which can be
+ used via the new "crossfade-ratio" property on the sink pads.
+
+- The avwait element has a new "end-timecode" property and posts
+ "avwait-status" element messages now whenever avwait starts or stops
+ passing through data (e.g. because target-timecode and end-timecode
+ respectively have been reached).
+
+ 
+- h265parse and h265parse will try harder to make upstream output the
+ same caps as downstream requires or prefers, thus avoiding
+ unnecessary conversion. The parsers also expose chroma format and
+ bit depth in the caps now.
+
+- The dtls elements now longer rely on or require the application to
+ run a GLib main loop that iterates the default main context
+ (GStreamer plugins should never rely on the application running a
+ GLib main loop).
+
+- openh264enc allows to change the encoding bitrate dynamically at
+ runtime now
+
+- nvdec is a new plugin for hardware-accelerated video decoding using
+ the NVIDIA NVDEC API (which replaces the old VDPAU API which is no
+ longer supported by NVIDIA)
+
+- The NVIDIA NVENC hardware-accelerated video encoders now support
+ dynamic bitrate and preset reconfiguration and support the I420
+ 4:2:0 video format. It's also possible to configure the gop size via
+ the new "gop-size" property.
+
+- The MPEG-TS muxer and demuxer (tsmux, tsdemux) now have support for
+ JPEG2000
+
+- openjpegdec and jpeg2000parse support 2-component images now (gray
+ with alpha), and jpeg2000parse has gained limited support for
+ conversion between JPEG2000 stream-formats. (JP2, J2C, JPC) and also
+ extracts more details such as colorimetry, interlace-mode,
+ field-order, multiview-mode and chroma siting.
+
+- The decklink plugin for Blackmagic capture and playback cards have
+ seen numerous improvements:
+
+- decklinkaudiosrc and decklinkvideosrc now put hardware reference
+ timestamp on buffers in form of GstReferenceTimestampMetas.
+ This can be useful to know on multi-channel cards which frames from
+ different channels were captured at the same time.
+
+- decklinkvideosink has gained support for Decklink hardware keying
+ with two new properties ("keyer-mode" and "keyer-level") to control
+ the built-in hardware keyer of Decklink cards.
+
+- decklinkaudiosink has been re-implemented around GstBaseSink instead
+ of the GstAudioBaseSink base class, since the Decklink APIs don't
+ fit very well with the GstAudioBaseSink APIs, which used to cause
+ various problems due to inaccuracies in the clock calculations.
+ Problems were audio drop-outs and A/V sync going wrong after
+ pausing/seeking.
+
+- support for more than 16 devices, without any artificial limit
+
+- work continued on the msdk plugin for Intel's Media SDK which
+ enables hardware-accelerated video encoding and decoding on Intel
+ graphics hardware on Windows or Linux. More tuning options were
+ added, and more pixel formats and video codecs are supported now.
+ The encoder now also handles force-key-unit events and can insert
+ frame-packing SEIs for side-by-side and top-bottom stereoscopic 3D
+ video.
+
+- dashdemux can now do adaptive trick play of certain types of DASH
+ streams, meaning it can do fast-forward/fast-rewind of normal (non-I
+ frame only) streams even at high speeds without saturating network
+ bandwidth or exceeding decoder capabilities. It will keep statistics
+ and skip keyframes or fragments as needed. See Sebastian's blog post
+ _DASH trick-mode playback in GStreamer_ for more details. It also
+ supports webvtt subtitle streams now and has seen improvements when
+ seeking in live streams.
+
+ 
+- kmssink has seen lots of fixes and improvements in this cycle,
+ including:
+
+- Raspberry Pi (vc4) and Xilinx DRM driver support
+
+- new "render-rectangle" property that can be used from the command
+ line as well as "display-width" and "display-height", and
+ "can-scale" properties
+
+- GstVideoCropMeta support
Plugin and library moves
-- this section will be filled in shortly
+MPEG-1 audio (mp1, mp2, mp3) decoders and encoders moved to -good
+
+Following the expiration of the last remaining mp3 patents in most
+jurisdictions, and the termination of the mp3 licensing program, as well
+as the decision by certain distros to officially start shipping full mp3
+decoding and encoding support, these plugins should now no longer be
+problematic for most distributors and have therefore been moved from
+-ugly and -bad to gst-plugins-good. Distributors can still disable these
+plugins if desired.
+
+In particular these are:
+
+- mpg123audiodec: an mp1/mp2/mp3 audio decoder using libmpg123
+- lamemp3enc: an mp3 encoder using LAME
+- twolamemp2enc: an mp2 encoder using TwoLAME
+
+GstAggregator moved from -bad to core
+
+GstAggregator has been moved from gst-plugins-bad to the base library in
+GStreamer and is now stable API.
+
+GstAggregator is a new base class for mixers and muxers that have to
+handle multiple input pads and aggregate streams into one output stream.
+It improves upon the existing GstCollectPads API in that it is a proper
+base class which was also designed with live streaming in mind.
+GstAggregator subclasses will operate in a mode with defined latency if
+any of the inputs are live streams. This ensures that the pipeline won't
+stall if any of the inputs stop producing data, and that the configured
+maximum latency is never exceeded.
+
+GstAudioAggregator, audiomixer and audiointerleave moved from -bad to -base
+
+GstAudioAggregator is a new base class for raw audio mixers and muxers
+and is based on GstAggregator (see above). It provides defined-latency
+mixing of raw audio inputs and ensures that the pipeline won't stall
+even if one of the input streams stops producing data.
+
+As part of the move to stabilise the API there were some last-minute API
+changes and clean-ups, but those should mostly affect internal elements.
+
+It is used by the audiomixer element, which is a replacement for
+'adder', which did not handle live inputs very well and did not align
+input streams according to running time. audiomixer should behave much
+better in that respect and generally behave as one would expected in
+most scenarios.
+
+Similarly, audiointerleave replaces the 'interleave' element which did
+not handle live inputs or non-aligned inputs very robustly.
+
+GstAudioAggregator and its subclases have gained support for input
+format conversion, which does not include sample rate conversion though
+as that would add additional latency. Furthermore, GAP events are now
+handled correctly.
+
+We hope to move the video equivalents (GstVideoAggregator and
+compositor) to -base in the next cycle, i.e. for 1.16.
+
+GStreamer OpenGL integration library and plugin moved from -bad to -base
+
+The GStreamer OpenGL integration library and opengl plugin have moved
+from gst-plugins-bad to -base and are now part of the stable API canon.
+Not all OpenGL elements have been moved; a few had to be left behind in
+gst-plugins-bad in the new openglmixers plugin, because they depend on
+the GstVideoAggregator base class which we were not able to move in this
+cycle. We hope to reunite these elements with the rest of their family
+for 1.16 though.
+
+This is quite a milestone, thanks to everyone who worked to make this
+happen!
+
+Qt QML and GTK plugins moved from -bad to -good
+
+The Qt QML-based qmlgl plugin has moved to -good and provides a
+qmlglsink video sink element as well as a qmlglsrc element. qmlglsink
+renders video into a QQuickItem, and qmlglsrc captures a window from a
+QML view and feeds it as video into a pipeline for further processing.
+Both elements leverage GStreamer's OpenGL integration. In addition to
+the move to -good the following features were added:
+
+- A proxy object is now used for thread-safe access to the QML widget
+ which prevents crashes in corner case scenarios: QML can destroy the
+ video widget at any time, so without this we might be left with a
+ dangling pointer.
+
+- EGL is now supported with the X11 backend, which works e.g. on
+ Freescale imx6
+
+The GTK+ plugin has also moved from -bad to -good. It includes gtksink
+and gtkglsink which both render video into a GtkWidget. gtksink uses
+Cairo for rendering the video, which will work everywhere in all
+scenarios but involves an extra memory copy, whereas gtkglsink fully
+leverages GStreamer's OpenGL integration, but might not work properly in
+all scenarios, e.g. where the OpenGL driver does not properly support
+multiple sharing contexts in different threads; on Linux Nouveau is
+known to be broken in this respect, whilst NVIDIA's proprietary drivers
+and most other drivers generally work fine, and the experience with
+Intel's driver seems to be fixed; some proprietary embedded Linux
+drivers don't work; macOS works).
+
+GstPhysMemoryAllocator interface moved from -bad to -base
+
+GstPhysMemoryAllocator is a marker interface for allocators with
+physical address backed memory.
Plugin removals
-- this section will be filled in shortly
-
-
-Miscellaneous API additions
+- the sunaudio plugin was removed, since it couldn't ever have been
+ built or used with GStreamer 1.0, but no one even noticed in all
+ these years.
-- this section will be filled in shortly
+- the schroedinger-based Dirac encoder/decoder plugin has been
+ removed, as there is no longer any upstream or anyone else
+ maintaining it. Seeing that it's quite a fringe codec it seemed best
+ to simply remove it.
-GstPlayer
+API removals
-- this section will be filled in shortly
+- some MPEG video parser API in the API unstable codecutils library in
+ gst-plugins-bad was removed after having been deprecated for 5
+ years.
Miscellaneous changes
-- this section will be filled in shortly
+- The video support library has gained support for a few new pixel
+ formats:
+- NV16_10LE32: 10-bit variant of NV16, packed into 32bit words (plus 2
+ bits padding)
+- NV12_10LE32: 10-bit variant of NV12, packed into 32bit words (plus 2
+ bits padding)
+- GRAY10_LE32: 10-bit grayscale, packed in 32bit words (plus 2 bits
+ padding)
+
+- decodebin, playbin and GstDiscoverer have seen stability
+ improvements in corner cases such as shutdown while still starting
+ up or shutdown in error cases (hat tip to the oss-fuzz project).
+
+- floating reference handling was inconsistent and has been cleaned up
+ across the board, including annotations. This solves various
+ long-standing memory leaks in language bindings, which e.g. often
+ caused elements and pads to be leaked.
+
+- major gobject-introspection annotation improvements for large parts
+ of the library API, including nullability of return types and
+ function parameters, correct types (e.g. strings vs. filenames),
+ ownership transfer, array length parameters, etc. This allows to use
+ bigger parts of the GStreamer API to be safely used from dynamic
+ language bindings (e.g. Python, Javascript) and allows static
+ bindings (e.g. C#, Rust, Vala) to autogenerate more API bindings
+ without manual intervention.
OpenGL integration
-- this section will be filled in shortly
+- The GStreamer OpenGL integration library has moved to
+ gst-plugins-base and is now part of our stable API.
+
+- new MESA3D GBM BACKEND. On devices with working libdrm support, it
+ is possible to use Mesa3D's GBM library to set up an EGL context
+ directly on top of KMS. This makes it possible to use the GStreamer
+ OpenGL elements without a windowing system if a libdrm- and
+ Mesa3D-supported GPU is present.
+
+- Prefer wayland display over X11: As most Wayland compositors support
+ XWayland, the X11 backend would get selected.
+
+- gldownload can export dmabufs now, and glupload will advertise
+ dmabuf as caps feature.
Tracing framework and debugging improvements
-- this section will be filled in shortly
+- NEW MEMORY RINGBUFFER BASED DEBUG LOGGER, useful for long-running
+ applications or to retrieve diagnostics when encountering an error.
+ The GStreamer debug logging system provides in-depth debug logging
+ about what is going on inside a pipeline. When enabled, debug logs
+ are usually written into a file, printed to the terminal, or handed
+ off to a log handler installed by the application. However, at
+ higher debug levels the volume of debug output quickly becomes
+ unmanageable, which poses a problem in disk-space or bandwidth
+ restricted environments or with long-running pipelines where a
+ problem might only manifest itself after multiple days. In those
+ situations, developers are usually only interested in the most
+ recent debug log output. The new in-memory ringbuffer logger makes
+ this easy: just installed it with gst_debug_add_ring_buffer_logger()
+ and retrieve logs with gst_debug_ring_buffer_logger_get_logs() when
+ needed. It is possible to limit the memory usage per thread and set
+ a timeout to determine how long messages are kept around. It was
+ always possible to implement this in the application with a custom
+ log handler of course, this just provides this functionality as part
+ of GStreamer.
+
+ 
+- 'fakevideosink is a null sink for video data that advertises
+ video-specific metas ane behaves like a video sink. See above for
+ more details.
+
+- gst_util_dump_buffer() prints the content of a buffer to stdout.
+
+- gst_pad_link_get_name() and gst_state_change_get_name() print pad
+ link return values and state change transition values as strings.
+
+- The LATENCY TRACER has seen a few improvements: trace records now
+ contain timestamps which is useful to plot things over time, and
+ downstream synchronisation time is now excluded from the measured
+ values.
+
+- Miniobject refcount tracing and logging was not entirley
+ thread-safe, there were duplicates or missing entries at times. This
+ has now been made reliable.
+
+- The netsim element, which can be used to simulate network jitter,
+ packet reordering and packet loss, received new features and
+ improvements: it can now also simulate network congestion using a
+ token bucket algorithm. This can be enabled via the "max-kbps"
+ property. Packet reordering can be disabled now via the
+ "allow-reordering" property: Reordering of packets is not very
+ common in networks, and the delay functions will always introduce
+ reordering if delay > packet-spacing, so by setting
+ "allow-reordering" to FALSE you guarantee that the packets are in
+ order, while at the same time introducing delay/jitter to them. By
+ using the new "delay-distribution" property the use can control how
+ the delay applied to delayed packets is distributed: This is either
+ the uniform distribution (as before) or the normal distribution; in
+ addition there is also the gamma distribution which simulates the
+ delay on wifi networks better.
Tools
-- this section will be filled in shortly
+- gst-inspect-1.0 now prints pad properties for elements that have pad
+ subclasses with special properties, such as compositor or
+ audiomixer. This only works for elements that use the newly-added
+ GstPadTemplate API API or the
+ gst_element_class_add_static_pad_template_with_gtype() convenience
+ function to tell GStreamer about the special pad subclass.
+
+- gst-launch-1.0 now generates a gstreamer pipeline diagram (.dot
+ file) whenever SIGHUP is sent to it on Linux/*nix systems.
+
+- gst-discoverer-1.0 can now analyse live streams such as rtsp:// URIs
GStreamer RTSP server
-- this section will be filled in shortly
+- Initial support for [RTSP protocol version
+ 2.0][rtsp2-lightning-talk] was added, which is to the best of our
+ knowledge the first RTSP 2.0 implementation ever!
+
+- ONVIF audio backchannel support. This is an extension specified by
+ ONVIF that allows RTSP clients (e.g. a control room operator) to
+ send audio back to the RTSP server (e.g. an IP camera).
+ Theoretically this could have been done also by using the RECORD
+ method of the RTSP protocol, but ONVIF chose not to do that, so the
+ backchannel is set up alongside the other streams. Format
+ negotiation needs to be done out of band, if needed. Use the new
+ ONVIF-specific subclasses GstRTSPOnvifServer and
+ GstRTSPOnvifMediaFactory to enable this functionality.
+
+ 
+- The internal server streaming pipeline is now dynamically
+ reconfigured on PLAY based on the transports needed. This means that
+ the server no longer adds the pipeline plumbing for all possible
+ transports from the start, but only if needed as needed. This
+ improves performance and memory footprint.
+
+- rtspclientsink has gained an "accept-certificate" signal for
+ manually checking a TLS certificate for validity.
+
+- Fix keep-alive/timeout issue for certain clients using TCP
+ interleave as transport who don't do keep-alive via some other
+ method such as periodic RTSP OPTION requests. We now put netaddress
+ metas on the packets from the TCP interleaved stream, so can map
+ RTCP packets to the right stream in the server and can handle them
+ properly.
+
+- Language bindings improvements: in general there were quite a few
+ improvements in the gobject-introspection annotations, but we also
+ extended the permissions API which was not usable from bindings
+ before.
+
+- Fix corner case issue where the wrong mount point was found when
+ there were multiple mount points with a common prefix.
GStreamer VAAPI
-- this section will be filled in shortly
+- this section will be filled in shortly {FIXME!}
GStreamer Editing Services and NLE
-- this section will be filled in shortly
+- this section will be filled in shortly {FIXME!}
GStreamer validate
-- this section will be filled in shortly
+- this section will be filled in shortly {FIXME!}
GStreamer Python Bindings
-- this section will be filled in shortly
+- this section will be filled in shortly {FIXME!}
Build and Dependencies
-- this section will be filled in shortly
+- the new WebRTC support in gst-plugins-bad depends on the GStreamer
+ elements that ship as part of libnice, and libnice version 1.1.14 is
+ required. Also the dtls and srtp plugins.
+
+- gst-plugins-bad no longer depends on the libschroedinger Dirac codec
+ library.
+
+- The srtp plugin can now also be built against libsrtp2.
+
+- some plugins and libraries have moved between modules, see the
+ _Plugin and_ _library moves_ section above, and their respective
+ dependencies have moved with them of course, e.g. the GStreamer
+ OpenGL integration support library and plugin is now in
+ gst-plugins-base, and mpg123, LAME and twoLAME based audio decoder
+ and encoder plugins are now in gst-plugins-good.
+
+- Unify static and dynamic plugin interface and remove plugin specific
+ static build option: Static and dynamic plugins now have the same
+ interface. The standard --enable-static/--enable-shared toggle is
+ sufficient. This allows building static and shared plugins from the
+ same object files, instead of having to build everything twice.
+
+- The default plugin entry point has changed. This will only affect
+ plugins that are recompiled against new GStreamer headers. Binary
+ plugins using the old entry point will continue to work. However,
+ plugins that are recompiled must have matching plugin names in
+ GST_PLUGIN_DEFINE and filenames, as the plugin entry point for
+ shared plugins is now deduced from the plugin filename. This means
+ you can no longer have a plugin called foo living in a file called
+ libfoobar.so or such, the plugin filename needs to match. This might
+ cause problems with some external third party plugin modules when
+ they get rebuilt against GStreamer 1.14.
+
+
+Note to packagers and distributors
+
+A number of libraries, APIs and plugins moved between modules and/or
+libraries in different modules between version 1.12.x and 1.14.x, see
+the _Plugin and_ _library moves_ section above. Some APIs have seen
+minor ABI changes in the course of moving them into the stable APIs
+section.
+
+This means that you should try to ensure that all major GStreamer
+modules are synced to the same major version (1.12 or 1.13/1.14) and can
+only be upgraded in lockstep, so that your users never end up with a mix
+of major versions on their system at the same time, as this may cause
+breakages.
+
+Also, plugins compiled against >= 1.14 headers will not load with
+GStreamer <= 1.12 owing to a new plugin entry point (but plugin binaries
+built against older GStreamer versions will continue to load with newer
+versions of GStreamer of course).
+
+There is also a small structure size related ABI breakage introduced in
+the gst-plugins-bad codecparsers library between version 1.13.90 and
+1.13.91. This should "only" affect gstreamer-vaapi, so anyone who ships
+the release candidates is advised to upgrade those two modules at the
+same time.
Platform-specific improvements
Android
-- this section will be filled in shortly
+- ahcsrc (Android camera source) does autofocus now
macOS and iOS
-- this section will be filled in shortly
+- this section will be filled in shortly {FIXME!}
Windows
-- this section will be filled in shortly
+- The GStreamer wasapi plugin was rewritten and should not only be
+ usable now, but in top shape and suitable for low-latency use cases.
+ The Windows Audio Session API (WASAPI) is Microsoft's most modern
+ method for talking with audio devices, and now that the wasapi
+ plugin is up to scratch it is preferred over the directsound plugin.
+ The ranks of the wasapisink and wasapisrc elements have been updated
+ to reflect this. Further improvements include:
+
+- support for more than 2 channels
+
+- a new "low-latency" property to enable low-latency operation (which
+ should always be safe to enable)
+
+- support for the AudioClient3 API which is only available on Windows
+ 10: in wasapisink this will be used automatically if available; in
+ wasapisrc it will have to be enabled explicitly via the
+ "use-audioclient3" property, as capturing audio with low latency and
+ without glitches seems to require setting the realtime priority of
+ the entire pipeline to "critical", which cannot be done from inside
+ the element, but has to be done in the application.
+
+- set realtime thread priority to avoid glitches
+
+- allow opening devices in exclusive mode, which provides much lower
+ latency compared to shared mode where WASAPI's engine period is
+ 10ms. This can be activated via the "exclusive" property.
+
+- There are now GstDeviceProvider implementations for the wasapi and
+ directsound plugins, so it's now possible to discover both audio
+ sources and audio sinks on Windows via the GstDeviceMonitor API
+
+- debug log timestamps are now higher granularity owing to
+ g_get_monotonic_time() now being used as fallback in
+ gst_utils_get_timestamp(). Before that, there would sometimes be
+ 10-20 lines of debug log output sporting the same timestamp.
Contributors
@@ -184,9 +1038,7 @@ suggestions or helped testing.
Bugs fixed in 1.14
-- this section will be filled in shortly
-
-More than 704 bugs have been fixed during the development of 1.14.
+More than 800 bugs have been fixed during the development of 1.14.
This list does not include issues that have been cherry-picked into the
stable 1.12 branch and fixed there as well, all fixes that ended up in
@@ -211,7 +1063,8 @@ the git 1.14 branch, which is a stable branch.
Known Issues
-- The webrtcdsp element is currently not shipped as part of the
+- The webrtcdsp element (which is unrelated to the newly-landed
+ GStreamer webrtc support) is currently not shipped as part of the
Windows binary packages due to a build system issue.
@@ -230,6 +1083,7 @@ several 1.15 pre-releases and the new 1.16 stable release in September.
------------------------------------------------------------------------
-_These release notes have been prepared by Tim-Philipp Müller._
+_These release notes have been prepared by Tim-Philipp Müller with_
+_contributions from Sebastian Dröge._
_License: CC BY-SA 4.0_