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GStreamer 1.16 Release Notes

GStreamer 1.16.0 was originally released on 19 April 2019.

See https://gstreamer.freedesktop.org/releases/1.16/ for the latest version of this document.

Last updated: Friday 19 April 2019, 00:00 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

  • New alternate fields interlace mode where each buffer carries a single field

  • WebM and Matroska ContentEncryption support in the Matroska demuxer

  • new WebKit WPE-based web browser source element

  • Video4Linux: HEVC encoding and decoding, JPEG encoding, and improved dmabuf import/export

  • Hardware-accelerated Nvidia video decoder gained support for VP8/VP9 decoding, whilst the encoder gained support for H.265/HEVC encoding.

  • Many improvements to the Intel Media SDK based hardware-accelerated video decoder and encoder plugin (msdk): dmabuf import/export for zero-copy integration with other components; VP9 decoding; 10-bit HEVC encoding; video post-processing (vpp) support including deinterlacing; and the video decoder now handles dynamic resolution changes.

  • The ASS/SSA subtitle overlay renderer can now handle multiple subtitles that overlap in time and will show them on screen simultaneously

  • The Meson build is now feature-complete (*) and it is now the recommended build system on all platforms. The Autotools build is scheduled to be removed in the next cycle.

  • The GStreamer Rust bindings and Rust plugins module are now officially part of upstream GStreamer.

  • The GStreamer Editing Services gained a gesdemux element that allows directly playing back serialized edit list with playbin or (uri)decodebin

  • Many performance improvements

Major new features and changes

Noteworthy new API

  • GstAggregator has a new "min-upstream-latency" property that forces a minimum aggregate latency for the input branches of an aggregator. This is useful for dynamic pipelines where branches with a higher latency might be added later after the pipeline is already up and running and where a change in the latency would be disruptive. This only applies to the case where at least one of the input branches is live though, it won't force the aggregator into live mode in the absence of any live inputs.

  • GstBaseSink gained a "processing-deadline" property and setter/getter API to configure a processing deadline for live pipelines. The processing deadline is the acceptable amount of time to process the media in a live pipeline before it reaches the sink. This is on top of the systemic latency that is normally reported by the latency query. This defaults to 20ms and should make pipelines such as v4l2src ! xvimagesink not claim that all frames are late in the QoS events. Ideally, this should replace the "max-lateness" property for most applications.

  • RTCP Extended Reports (XR) parsing according to RFC 3611: Loss/Duplicate RLE, Packet Receipt Times, Receiver Reference Time, Delay since the last Receiver (DLRR), Statistics Summary, and VoIP Metrics reports. This only provides the ability to parse such packets, generation of XR packets is not supported yet and XR packets are not automatically parsed by rtpbin / rtpsession but must be actively handled by the application.

  • a new mode for interlaced video was added where each buffer carries a single field of interlaced video, with buffer flags indicating whether the field is the top field or bottom field. Top and bottom fields are expected to alternate in this mode. Caps for this interlace mode must also carry a format:Interlaced caps feature to ensure backwards compatibility.

  • The video library has gained support for three new raw pixel formats:

    • Y410: packed 4:4:4 YUV, 10 bits per channel
    • Y210: packed 4:2:2 YUV, 10 bits per channel
    • NV12_10LE40: fully-packed 10-bit variant of NV12_10LE32, i.e. without the padding bits
  • GstRTPSourceMeta is a new meta that can be used to transport information about the origin of depayloaded or decoded RTP buffers, e.g. when mixing audio from multiple sources into a single stream. A new "source-info" property on the RTP depayloader base class determines whether depayloaders should put this meta on outgoing buffers. Similarly, the same property on RTP payloaders determines whether they should use the information from this meta to construct the CSRCs list on outgoing RTP buffers.

  • gst_sdp_message_from_text() is a convenience constructor to parse SDPs from a string which is particularly useful for language bindings.

Support for Planar (Non-Interleaved) Raw Audio

Raw audio samples are usually passed around in interleaved form in GStreamer, which means that if there are multiple audio channels the samples for each channel are interleaved in memory, e.g. |LEFT|RIGHT|LEFT|RIGHT|LEFT|RIGHT| for stereo audio. A non-interleaved or planar arrangement in memory would look like |LEFT|LEFT|LEFT|RIGHT|RIGHT|RIGHT| instead, possibly with |LEFT|LEFT|LEFT| and |RIGHT|RIGHT|RIGHT| residing in separate memory chunks or separated by some padding.

GStreamer has always had signalling for non-interleaved audio since version 1.0, but it was never actually properly implemented in any elements. audioconvert would advertise support for it, but wasn't actually able to handle it correctly.

With this release we now have full support for non-interleaved audio as well, which means more efficient integration with external APIs that handle audio this way, but also more efficient processing of certain operations like interleaving multiple 1-channel streams into a multi-channel stream which can be done without memory copies now.

New API to support this has been added to the GStreamer Audio support library: There is now a new GstAudioMeta which describes how data is laid out inside the buffer, and buffers with non-interleaved audio must always carry this meta. To access the non-interleaved audio samples you must map such buffers with gst_audio_buffer_map() which works much like gst_buffer_map() or gst_video_frame_map() in that it will populate a little GstAudioBuffer helper structure passed to it with the number of samples, the number of planes and pointers to the start of each plane in memory. This function can also be used to map interleaved audio buffers in which case there will be only one plane of interleaved samples.

Of course support for this has also been implemented in the various audio helper and conversion APIs, base classes, and in elements such as audioconvert, audioresample, audiotestsrc, audiorate.

Support for Closed Captions and Other Ancillary Data in Video

The video support library has gained support for detecting and extracting Ancillary Data from videos as per the SMPTE S291M specification, including:

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

  • line21encoder, line21decoder: inject/extract line21 closed captions to/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, h264parse extracts Closed Captions from MPEG-2/H.264 video streams

  • avviddec, avvidenc, x264enc got support for extracting/injecting Closed Captions

  • 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, line21encoder, 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

  • srtpdec, srtpenc: add support for AES-GCM and also add support for it in gst-rtsp-server and rtspsrc.

  • 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 and supports the XDG-Shell protocol.

  • decklinkvideosink, decklinkvideosrc support selecting between half/full duplex

  • The vulkan plugin gained support for macOS and iOS via MoltenVK in addition to the existing support for X11 and Wayland

  • imagefreeze has a new num-buffers property to limit the number of buffers that are produced and to send an EOS event afterwards

  • webrtcbin has a new, introspectable get-transceiver signal in addition to the old get-transceivers signal that couldn't be used from bindings

  • Support for per-element latency information was added to the latency tracer

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

  • GstDiscoverer has support for caching the results of discovery in the default cache directory. This can be enabled with the use-cache property and is disabled by default.

  • GstMeta that are attached to GstBuffers are now always stored in the order in which they were added.

  • Additional support for signalling ONVIF specific features were added: the SEEK event can store a trickmode-interval now and support for the Rate-Control and Frames RTSP headers was added to the RTSP library.

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.

  • The OpenGL library no longer restores the OpenGL viewport. This is a performance optimization to not require performing multiple expensive glGet*() function calls per frame. This affects any application or plugin use of the following functions and objects:

    • glcolorconvert library object (not the element)
    • glviewconvert library object (not the element)
    • gst_gl_framebuffer_draw_to_texture()
    • custom GstGLWindow implementations

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:
      gst-print example

  • 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.

Tools

  • gst-inspect-1.0 has coloured output now and will automatically use a pager if the output does not fit on a page. This only works in a UNIX environment and if the output is not piped, and on Windows 10 build 16257 or newer. If you don't like the colours you can disable them by setting the GST_INSPECT_NO_COLORS=1 environment variable or passing the --no-color command line option.

GStreamer RTSP server

  • Improved backlog handling when using TCP interleaved for data transport. Before there was a fixed maximum size for backlog messages, which was prone to deadlocks and made it difficult to control memory usage with the watch backlog. The RTSP server now limits queued TCP data messages to one per stream, moving queuing of the data into the pipeline and leaving the RTSP connection responsive to RTSP messages in both directions, preventing all those problems.

  • Initial ULP Forward Error Correction support in rtspclientsink and for RECORD mode in the server.

  • API to explicitly enable retransmission requests (RTX)

  • Lots of multicast-related fixes

  • rtsp-auth: Add support for parsing .htdigest files

GStreamer VAAPI

  • Support Wayland's display for context sharing, so the application can pass its own wl_display in order to be used for the VAAPI display creation.

  • A lot of work to support new Intel hardware using media-driver as VA backend.

  • For non-x86 devices, VAAPI display can instantiate, through DRM, with no PCI bus. This enables the usage of libva-v4l2-request driver.

  • Added support for XDG-shell protocol as wl_shell replacement which is currently deprecated. This change add as dependency wayland-protocol.

  • GstVaapiFilter, GstVaapiWindow, and GstVaapiDecoder classes now inherit from GstObject, gaining all the GStreamer's instrumentation support.

  • The metadata now specifies the plugin as Hardware class.

  • H264 decoder is more stable with problematic streams.

  • In H265 decoder added support for profiles main-422-10 (P010_10LE), main-444 (AYUV) and main-444-10 (Y410)

  • JPEG decoder handles dynamic resolution changes.

  • More specification adherence in H264 and H265 encoders.

GStreamer OMX

  • Add support of NV16 format to video encoders input.

  • Video decoders now handle the ALLOCATION query to tell upstream about the number of buffers they require. Video encoders will also use this query to adjust their number of allocated buffers preventing starvation when using dynamic buffer mode.

  • The OMX_PERFORMANCE debug category has been renamed to OMX_API_TRACE and can now be used to track a widder variety of interactions between OMX and GStreamer.

  • Video encoders will now detect frame rate only changes and will inform OMX about it rather than doing a full format reset.

  • Various Zynq UltraScale+ specific improvements:

    • Video encoders are now able to import dmabuf from upstream.
    • Support for HEVC range extension profiles and more AVC profiles.
    • We can now request video encoders to generate an IDR using the force key unit event.

GStreamer Editing Services and NLE

  • Added a gesdemux element, it is an auto pluggable element that allows decoding edit list like files supported by GES

  • Added gessrc which wraps a GESTimeline as a standard source element (implementing the ges protocol handler)

  • Added basic support for videorate::rate property potentially allowing changing playback speed

  • Layer priority is now fully automatic and they should be moved with the new ges_timeline_move_layer method, ges_layer_set_priority is now deprecated.

  • Added a ges_timeline_element_get_layer_priority so we can simply get all information about GESTimelineElement position in the timeline

  • GESVideoSource now auto orientates the images if it is defined in a meta (overridable).

  • Added some PyGObject overrides to make the API more pythonic

  • The threading model has been made more explicit with safe guard to make sure not thread safe APIs are not used from the wrong threads. It is also now possible to properly handle in what thread the API should be used.

  • Optimized GESClip and GESTrackElement creation

  • Added a way to compile out the old, unused and deprecated GESPitiviFormatter

  • Re implemented the timeline editing API making it faster and making the code much more maintainable

  • Simplified usage of nlecomposition outside GES by removing quirks in it API usage and removing the need to treat it specially from an application perspective.

  • ges-launch-1.0:

    • Added support to add titles to the timeline
    • Enhance the help auto generating it from the code
  • Deprecate ges_timeline_load_from_uri as loading the timeline should be done through a project now

  • Many leaks have been plugged and the unit testsuite is now "leak free"

GStreamer validate

  • Added an action type to verify the checksum of the sink last-sample

  • Added an include keyword to validate scenarios

  • Added the notion of variable in scenarios, with the set-vars keyword

  • Started adding support for "performance" like tests by allowing to define the number of dropped buffers or the minimum buffer frequency on a specific pad

  • Added a validateflow plugin which allows defining the data flow to be seen on a particular pad and verifying that following runs match the expectations

  • Added support for appsrc based test definition so we can instrument the data pushed into the pipeline from scenarios

  • Added a mockdecryptor allowing adding tests with on encrypted files, the element will potentially be instrumented with a validate scenario

  • gst-validate-launcher:

    • Cleaned up output

    • Changed the default for "muting" tests as user doesn't expect hundreds of windows to show up when running the testsuite

    • Fixed the outputted xunit files to be compatible with GitLab

    • Added support to run tests on media files in push mode (using pushfile://)

    • Added support for running inside gst-build

    • Added support for running ssim tests on rendered files

    • Added a way to simply define tests on pipelines through a simple .json file

    • Added a python app to easily run python testsuite reusing all the launcher features

    • Added flatpak knowledge so we can print backtrace even when running from within flatpak

    • Added a way to automatically generated "known issues" suppressions lines

    • Added a way to rerun tests to check if they are flaky and added a way to tolerate tests known to be flaky

    • Add a way to output html log files

GStreamer Python Bindings

  • add binding for gst_pad_set_caps()

  • pygobject dependency requirement was bumped to >= 3.8

  • new audiotestsrc, audioplot, and mixer plugin examples, and a dynamic pipeline example

GStreamer C# Bindings

  • bindings for the GstWebRTC library

GStreamer Rust Bindings

The GStreamer Rust bindings are now officially part of the GStreamer project and are also maintained in the GStreamer GitLab.

The releases will generally not be synchronized with the releases of other GStreamer parts due to dependencies on other projects.

Also unlike the other GStreamer libraries, the bindings will not commit to full API stability but instead will follow the approach that is generally taken by Rust projects, e.g.:

1) 0.12.X will be completely API compatible with all other 0.12.Y versions. 2) 0.12.X+1 will contain bugfixes and compatible new feature additions. 3) 0.13.0 will not be backwards compatible with 0.12.X but projects will be able to stay at 0.12.X without any problems as long as they don't need newer features.

The current stable release is 0.12.2 and the next release series will be 0.13, probably around March 2019.

At this point the bindings cover most of GStreamer core (except for most notably GstAllocator and GstMemory), and most parts of the app, audio, base, check, editing-services, gl, net. pbutils, player, rtsp, rtsp-server, sdp, video and webrtc libraries.

Also included is support for creating subclasses of the following types and writing GStreamer plugins:

  • gst::Element
  • gst::Bin and gst::Pipeline
  • gst::URIHandler and gst::ChildProxy
  • gst::Pad, gst::GhostPad
  • gst_base::Aggregator and gst_base::AggregatorPad
  • gst_base::BaseSrc and gst_base::BaseSink
  • gst_base::BaseTransform

Changes to 0.12.X since 0.12.0

Fixed

  • PTP clock constructor actually creates a PTP instead of NTP clock

Added

  • Bindings for GStreamer Editing Services
  • Bindings for GStreamer Check testing library
  • Bindings for the encoding profile API (encodebin)

  • VideoFrame, VideoInfo, AudioInfo, StructureRef implements Send and Sync now

  • VideoFrame has a function to get the raw FFI pointer
  • From impls from the Error/Success enums to the combined enums like FlowReturn
  • Bin-to-dot file functions were added to the Bin trait
  • gst_base::Adapter implements SendUnique now
  • More complete bindings for the gst_video::VideoOverlay interface, especially gst_video::is_video_overlay_prepare_window_handle_message()

Changed

  • All references were updated from GitHub to freedesktop.org GitLab
  • Fix various links in the README.md
  • Link to the correct location for the documentation
  • Remove GitLab badge as that only works with gitlab.com currently

Changes in git master for 0.13

Fixed

  • gst::tag::Album is the album tag now instead of artist sortname

Added

  • Subclassing infrastructure was moved directly into the bindings, making the gst-plugin crate deprecated. This involves many API changes but generally cleans up code and makes it more flexible. Take a look at the gst-plugins-rs crate for various examples.

  • Bindings for CapsFeatures and Meta

  • Bindings for ParentBufferMeta,VideoMetaandVideoOverlayCompositionMeta`
  • Bindings for VideoOverlayComposition and VideoOverlayRectangle
  • Bindings for VideoTimeCode

  • UniqueFlowCombiner and UniqueAdapter wrappers that make use of the Rust compile-time mutability checks and expose more API in a safe way, and as a side-effect implement Sync and Send now

  • More complete bindings for Allocation Query

  • pbutils functions for codec descriptions
  • TagList::iter() for iterating over all tags while getting a single value per tag. The old ::iter_tag_list() function was renamed to ::iter_generic() and still provides access to each value for a tag
  • Bus::iter() and Bus::iter_timed() iterators around the corresponding ::pop\*() functions

  • serde serialization of Value can also handle Buffer now

  • Extensive comments to all examples with explanations

  • Transmuxing example showing how to use typefind, multiqueue and dynamic pads
  • basic-tutorial-12 was ported and added

Changed

  • Rust 1.31 is the minimum supported Rust version now
  • Update to latest gir code generator and glib bindings

  • Functions returning e.g. gst::FlowReturn or other "combined" enums were changed to return split enums like Result<gst::FlowSuccess, gst::FlowError> to allow usage of the standard Rust error handling.

  • MiniObject subclasses are now newtype wrappers around the underlying GstRc<FooRef> wrapper. This does not change the API in any breaking way for the current usages, but allows MiniObjects to also be implemented in other crates and makes sure rustdoc places the documentation in the right places.

  • BinExt extension trait was renamed to GstBinExt to prevent conflicts with gtk::Bin if both are imported

  • Buffer::from_slice() can't possible return None

  • Various clippy warnings

GStreamer Rust Plugins

Like the GStreamer Rust bindings, the Rust plugins are now officially part of the GStreamer project and are also maintained in the GStreamer GitLab.

In the 0.3.x versions this contained infrastructure for writing GStreamer plugins in Rust, and a set of plugins.

In git master that infrastructure was moved to the GLib and GStreamer bindings directly, together with many other improvements that were made possible by this, so the gst-plugins-rs repository only contains GStreamer elements now.

Elements included are:

  • Tutorials plugin: identity, rgb2gray and sinesrc with extensive comments

  • rsaudioecho, a port of the audiofx element

  • rsfilesrc, rsfilesink

  • rsflvdemux, a FLV demuxer. Not feature-equivalent with flvdemux yet

  • threadshare plugin: ts-appsrc, ts-proxysrc/sink, ts-queue, ts-udpsrc and ts-tcpclientsrc elements that use a fixed number of threads and share them between instances. For more background about these elements see Sebastian's talk "When adding more threads adds more problems - Thread-sharing between elements in GStreamer" at the GStreamer Conference 2017.

  • rshttpsrc, a HTTP source around the hyper/reqwest Rust libraries. Not feature-equivalent with souphttpsrc yet.

  • togglerecord, an element that allows to start/stop recording at any time and keeps all audio/video streams in sync.

  • mccparse and mccenc, parsers and encoders for the MCC closed caption file format.

Changes to 0.3.X since 0.3.0

  • All references were updated from GitHub to freedesktop.org GitLab
  • Fix various links in the README.md
  • Link to the correct location for the documentation

Changes in git master for 0.4

  • togglerecord: Switch to parking_lot crate for mutexes/condition variables for lower overhead
  • Merge threadshare plugin here
  • New closedcaption plugin with mccparse and mccenc elements
  • New identity element for the tutorials plugin

  • Register plugins statically in tests instead of relying on the plugin loader to find the shared library in a specific place

  • Update to the latest API changes in the GLib and GStreamer bindings

  • Update to the latest versions of all crates

Build and Dependencies

  • The Meson build system build is now feature-complete (*) and it is now the recommended build system on all platforms and also used by Cerbero to build GStreamer on all platforms. The Autotools build is scheduled to be removed in the next cycle. Developers who currently use gst-uninstalled should move to gst-build. The build option naming has been cleaned up and made consistent and there are now feature options to enable/disable plugins and various other features on a case-by-case basis. (*) with the exception of plugin docs which will be handled differently in future

  • Symbol export in libraries is now controlled via explicit exports using symbol visibility or export defines where supported, to ensure consistency across all platforms. This also allows libraries to have exports that vary based on detected platform features and configure options as is the case with the GStreamer OpenGL integration library for example. A few symbols that had been exported by accident in earlier versions may no longer be exported. These symbols will not have had declarations in any public header files then though and would not have been usable.

  • The GStreamer FFmpeg wrapper plugin (gst-libav) now depends on FFmpeg 4.x and uses the new FFmpeg 4.x API and stopped relying on ancient API that was removed with the FFmpeg 4.x release. This means that it is no longer possible to build this module against an older system-provided FFmpeg 3.x version. Use the internal FFmpeg 4.x copy instead if you build using autotools, or use gst-libav 1.14.x instead which targets the FFmpeg 3.x API and should work fine in combination with a newer GStreamer. It's difficult for us to support both old and new FFmpeg APIs at the same time, apologies for any inconvenience caused.

  • Hardware-accelerated Nvidia video encoder/decoder plugins nvdec and nvenc can be built against CUDA Toolkit versions 9 and 10.0 now. The dynlink interface has been dropped since it's deprecated in 10.0.

  • The (optional) OpenCV requirement has been bumped to >= 3.0.0 and the plugin can also be built against OpenCV 4.x now.

  • New sctp plugin based on usrsctp (for WebRTC data channels)

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.

  • When building on Windows, Cerbero can now build GStreamer recipes and core dependencies such as glib with Visual Studio. This is controlled by the visualstudio variant. Visual Studio 2015, 2017, and 2019 are supported. Currently, only 64-bit x86 is supported due to a known bug which will be fixed for the next release.

  • 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][gio-gnutls-static-link-example] 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][gio-gnutls-static-link-example] for the necessary change in the examples.

[gio-gnutls-static-link-example]: insert gst-examples/tutorial commit (FIXME)

Windows

  • The webrtcdsp element is shipped again as part of the Windows binary packages, the build system issue has been resolved.

  • 'Inconsistent DLL linkage' warnings when building with MSVC have been fixed

  • Hardware-accelerated Nvidia video encoder/decoder plugins nvdec and nvenc build on Windows now, also with MSVC and using Meson.

  • The ksvideosrc camera capture plugin supports 16-bit grayscale video now

  • 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

Aaron Boxer, 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, Charlie Turner, Christoph Reiter, Corentin Noël, Daeseok Youn, Damian Vicino, Dan Kegel, Daniel Drake, Daniel Klamt, Danilo Spinella, Dardo D Kleiner, David Ing, David Svensson Fors, Devarsh Thakkar, Dimitrios Katsaros, Edward Hervey, Emilio Pozuelo Monfort, Enrique Ocaña González, Erlend Eriksen, 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, Ilya Smelykh, Ingo Randolf, Iñigo Huguet, Jakub Adam, James Stevenson, Jan Alexander Steffens, Jan Schmidt, Jerome Laheurte, Jimmy Ohn, Joakim Johansson, Jochen Henneberg, Johan Bjäreholt, John-Mark Bell, John Bassett, John Nikolaides, Jonathan Karlsson, Jonny Lamb, Jordan Petridis, Josep Torra, Joshua M. Doe, Jos van Egmond, Juan Navarro, Julian Bouzas, Jun Xie, Junyan He, Justin Kim, Kai Kang, Kim Tae Soo, Kirill Marinushkin, Kyrylo Polezhaiev, Lars Petter Endresen, Linus Svensson, Louis-Francis Ratté-Boulianne, Lucas Stach, Luis de Bethencourt, Luz Paz, Lyon Wang, Maciej Wolny, Marc-André Lureau, Marc Leeman, Marco Trevisan (Treviño), 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, Pat DeSantis, Patricia Muscalu, Patrick Radizi, Patrik Nilsson, Paul Kocialkowski, Per Forlin, Peter Körner, Peter Seiderer, Petr Kulhavy, Philippe Normand, Philippe Renon, Philipp Zabel, Pierre Labastie, Piotr Drąg, Roland Jon, Roman Sivriver, Roman Shpuntov, Rosen Penev, Russel Winder, Sam Gigliotti, Santiago Carot-Nemesio, Sean-Der, Sebastian Dröge, Seungha Yang, Shi Yan, 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, Tobias Ronge, 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.

Stable 1.16 branch

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.

1.16.0

1.16.0 was released on 19 April 2019.

Known Issues

  • possibly breaking/incompatible changes to properties of wrapped FFmpeg decoders and encoders (see above).

  • The way that GIO modules are named has changed due to upstream GLib natively adding support for loading static GIO modules. This means that any GStreamer application using gnutls for SSL/TLS on the Android or iOS platforms (or any other setup using static libraries) will fail to link looking for the g_io_module_gnutls_load_static() function. The new function name is now g_io_gnutls_load(gpointer data). See Android/iOS sections above for further details.

Schedule for 1.18

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 1.18 development cycle is yet to be confirmed, but it is possible that the next cycle will be a short one in which case feature freeze would be perhaps around August 2019 with a new 1.18 stable release in September.

1.18 will be backwards-compatible to the stable 1.16, 1.14, 1.12, 1.10, 1.8, 1.6, 1.4, 1.2 and 1.0 release series.


These release notes have been prepared by Tim-Philipp Müller with contributions from Sebastian Dröge, Guillaume Desmottes, Matthew Waters, * *Thibault Saunier, and Víctor Manuel Jáquez Leal.

License: CC BY-SA 4.0


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