GStreamer Changelog

What's new in GStreamer 1.16.0

Apr 23, 2019

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 (with the exception of plugin docs) 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

New in GStreamer 1.14.1 (May 18, 2018)

New in GStreamer 1.14.0 (Mar 20, 2018)

New in GStreamer 1.12.0 (May 4, 2017)

Highlights:
new msdk plugin for Intel's Media SDK for hardware-accelerated video encoding and decoding on Intel graphics hardware on Windows or Linux.
x264enc can now use multiple x264 library versions compiled for different bit depths at runtime, to transparently provide support for multiple bit depths.
videoscale and videoconvert now support multi-threaded scaling and conversion, which is particularly useful with higher resolution video.
h264parse will now automatically insert AU delimiters if needed when outputting byte-stream format, which improves standard compliance and is needed in particular for HLS playback on iOS/macOS.
rtpbin has acquired bundle support for incoming streams
Major new features and changes:
Noteworthy new API:
The video library gained support for a number of new video formats:GBR_12LE, GBR_12BE, GBRA_12LE, GBRA_12BE (planar 4:4:4 RGB/RGBA, 12 bits per channel)GBRA_10LE, GBRA_10BE (planar 4:4:4:4 RGBA, 10 bits per channel)GBRA (planar 4:4:4:4 ARGB, 8 bits per channel)I420_12BE, I420_12LE (planar 4:2:0 YUV, 12 bits per channel)I422_12BE,I422_12LE (planar 4:2:2 YUV, 12 bits per channel)Y444_12BE, Y444_12LE (planar 4:4:4 YUV, 12 bits per channel)VYUY (another packed 4:2:2 YUV format)
The high-level GstPlayer API was extended with functions for taking video snapshots and enabling accurate seeking. It can optionally also use the still-experimental playbin3 element now.
New Elements:
msdk: new plugin for Intel's Media SDK for hardware-accelerated video encoding and decoding on Intel graphics hardware on Windows or Linux. This includes an H.264 encoder/decoder (msdkh264dec, msdkh264enc), an H.265 encoder/decoder (msdkh265dec, msdkh265enc), an MJPEG encoder/encoder (msdkmjpegdec, msdkmjpegenc), an MPEG-2 video encoder (msdkmpeg2enc) and a VP8 encoder (msdkvp8enc).
iqa is a new Image Quality Assessment plugin based on DSSIM, similar to the old (unported) videomeasure element.
The faceoverlay element, which allows you to overlay SVG graphics over a detected face in a video stream, has been ported from 0.10.
our ffmpeg wrapper plugin now exposes/maps the ffmpeg Opus audio decoder (avdec_opus) as well as the GoPro CineForm HD / CFHD decoder (avdec_cfhd), and also a parser/writer for the IVF format (avdemux_ivf and avmux_ivf).
audiobuffersplit is a new element that splits raw audio buffers into equal-sized buffers
audiomixmatrix is a new element that mixes N:M audio channels according to a configured mix matrix.
The timecodewait element got renamed to avwait and can operate in different modes now.
The opencv video processing plugin has gained a new dewarp element that dewarps fisheye images.
ttml is a new plugin for parsing and rendering subtitles in Timed Text Markup Language (TTML) format. For the time being these elements will not be autoplugged during media playback however, unless the GST_TTML_AUTOPLUG=1 environment variable is set. Only the EBU-TT-D profile is supported at this point.
New element features and additions:
x264enc can now use multiple x264 library versions compiled for different bit depths at runtime, to transparently provide support for multiple bit depths. A new configure parameter --with-x264-libraries has been added to specify additional paths to look for additional x264 libraries to load. Background is that the libx264 library is always compile for one specific bit depth and the x264enc element would simply support the depth supported by the underlying library. Now we can support multiple depths.
x264enc also picks up the interlacing mode automatically from the input caps now and passed interlacing/TFF information correctly to the library.
videoscale and videoconvert now support multi-threaded scaling and conversion, which is particularly useful with higher resolution video. This has to be enabled explicitly via the "n-threads" property.
videorate's new "rate" property lets you set a speed factor on the output stream
splitmuxsink's buffer collection and scheduling was rewritten to make processing and splitting deterministic; before it was possible for a buffer to end up in a different file chunk in different runs. splitmuxsink also gained a new "format-location-full" signal that works just like the existing "format-location" signal only that it is also passed the primary stream's first buffer as argument, so that it is possible to construct the file name based on metadata such as the buffer timestamp or any GstMeta attached to the buffer. The new "max-size-timecode" property allows for timecode-based splitting. splitmuxsink will now also automatically start a new file if the input caps change in an incompatible way.
fakesink has a new "drop-out-of-segment" property to not drop out-of-segment buffers, which is useful for debugging purposes.
identity gained a "ts-offset" property.
both fakesink and identity now also print what kind of metas are attached to buffers when printing buffer details via the "last-message" property used by gst-launch-1.0 -v.
multiqueue: made "min-interleave-time" a configurable property.
video nerds will be thrilled to know that videotestsrc's snow is now deterministic. videotestsrc also gained some new properties to make the ball pattern based on system time, and invert colours each second ("animation-mode", "motion", and "flip" properties).
oggdemux reverse playback should work again now. You're welcome.
playbin3 and urisourcebin now have buffering enabled by default, and buffering message aggregation was fixed.
tcpclientsrc now has a "timeout" property
appsink has gained support for buffer lists. For backwards compatibility reasons users need to enable this explicitly with gst_app_sink_set_buffer_list_support(), however. Once activated, a pulled GstSample can contain either a buffer list or a single buffer.
splitmuxsrc reverse playback was fixed and handling of sparse streams, such as subtitle tracks or metadata tracks, was improved.
matroskamux has acquired support for muxing G722 audio; it also marks all buffers as keyframes now when streaming only audio, so that tcpserversink will behave properly with audio-only streams.
qtmux gained support for ProRes 4444 XQ, HEVC/H.265 and CineForm (GoPro) formats, and generally writes more video stream-related metadata into the track headers. It is also allows configuration of the maximum interleave size in bytes and time now. For fragmented mp4 we always write the tfdt atom now as required by the DASH spec.
qtdemux supports FLAC, xvid, mp2, S16L and CineForm (GoPro) tracks now, and generally tries harder to extract more video-related information from track headers, such as colorimetry or interlacing details. It also received a couple of fixes for the scenario where upstream operates in TIME format and feeds chunks to qtdemux (e.g. DASH or MSE).
audioecho has two new properties to apply a delay only to certain channels to create a surround effect, rather than an echo on all channels. This is useful when upmixing from stereo, for example. The "surround-delay" property enables this, and the "surround-mask" property controls which channels are considered surround sound channels in this case.
webrtcdsp gained various new properties for gain control and also exposes voice activity detection now, in which case it will post "voice-activity" messages on the bus whenever the voice detection status changes.
The decklink capture elements for Blackmagic Decklink cards have seen a number of improvements:
decklinkvideosrc will post a warning message on "no signal" and an info message when the signal lock has been (re)acquired. There is also a new read-only "signal" property that can be used to query the signal lock status. The GAP flag will be set on buffers that are captured without a signal lock. The new drop-no-signal-frames will make decklinkvideosrc drop all buffers that have been captured without an input signal. The "skip-first-time" property will make the source drop the first few buffers, which is handy since some devices will at first output buffers with the wrong resolution before they manage to figure out the right input format and decide on the actual output caps.
decklinkaudiosrc supports more than just 2 audio channels now.
The capture sources no longer use the "hardware" timestamps which turn out to be useless and instead just use the pipeline clock directly.
srtpdec now also has a readonly "stats" property, just like srtpenc.
rtpbin gained RTP bundle support, as used by e.g. WebRTC. The first rtpsession will have a rtpssrcdemux element inside splitting the streams based on their SSRC and potentially dispatch to a different rtpsession. Because retransmission SSRCs need to be merged with the corresponding media stream the ::on-bundled-ssrc signal is emitted on rtpbin so that the application can find out to which session the SSRC belongs.
rtprtxqueue gained two new properties exposing retransmission statistics ("requests" and "fulfilled-requests")
kmssink will now use the preferred mode for the monitor and render to the base plane if nothing else has set a mode yet. This can also be done forcibly in any case via the new "force-modesetting" property. Furthermore, kmssink now allows only the supported connector resolutions as input caps in order to avoid scaling or positioning of the input stream, as kmssink can't know whether scaling or positioning would be more appropriate for the use case at hand.
waylandsink can now take DMAbuf buffers as input in the presence of a compatible Wayland compositor. This enables zero-copy transfer from a decoder or source that outputs DMAbuf. It will also set surface opacity hint to allow better rendering optimization in the compositor.
udpsrc can be bound to more than one interface when joining a multicast group, this is done by giving a comma separate list of interfaces such as multicast-iface="eth0,eth1".
Plugin moves:
dataurisrc moved from gst-plugins-bad to core
The rawparse plugin containing the rawaudioparse and rawvideoparse elements moved from gst-plugins-bad to gst-plugins-base. These elements supersede the old videoparse and audioparse elements. They work the same, with just some minor API changes. The old legacy elements still exist in gst-plugins-bad, but may be removed at some point in the future.
timecodestamper is an element that attaches time codes to video buffers in form of GstVideoTimeCodeMetas. It had a "clock-source" property which has now been removed because it was fairly useless in practice. It gained some new properties however: the "first-timecode" property can be used to set the inital timecode; alternatively "first-timecode-to-now" can be set, and then the current system time at the time the first buffer arrives is used as base time for the time codes.
Plugin removals:
The mad mp1/mp2/mp3 decoder plugin was removed from gst-plugins-ugly, as libmad is GPL licensed, has been unmaintained for a very long time, and there are better alternatives available. Use the mpg123audiodec element from the mpg123 plugin in gst-plugins-ugly instead, or avdec_mp3 from the gst-libav module which wraps the ffmpeg library. We expect that we will be able to move mp3 decoding to gst-plugins-good in the next cycle seeing that most patents around mp3 have expired recently or are about to expire.
The mimic plugin was removed from gst-plugins-bad. It contained a decoder and encoder for a video codec used by MSN messenger many many years ago (in a galaxy far far away). The underlying library is unmaintained and no one really needs to use this codec any more. Recorded videos can still be played back with the MIMIC decoder in gst-libav.
Miscellaneous API additions:
Request pad name templates passed to gst_element_request_pad() may now contain multiple specifiers, such as e.g. src_%u_%u.
gst_buffer_iterate_meta_filtered() is a variant of gst_buffer_iterate_meta() that only returns metas of the requested type and skips all other metas.
gst_pad_task_get_state() gets the current state of a task in a thread-safe way.
gst_uri_get_media_fragment_table() provides the media fragments of an URI as a table of key=value pairs.
gst_print(), gst_println(), gst_printerr(), and gst_printerrln() can be used to print to stdout or stderr. These functions are similar to g_print() and g_printerr() but they also support all the additional format specifiers provided by the GStreamer logging system, such as e.g. GST_PTR_FORMAT.
a GstParamSpecArray has been added, for elements who want to have array type properties, such as the audiomixmatrix element for example. There are also two new functions to set and get properties of this type from bindings:gst_util_set_object_array()gst_util_get_object_array()
various helper functions have been added to make it easier to set or get GstStructure fields containing caps-style array or list fields from language bindings (which usually support GValueArray but don't know about the GStreamer specific fundamental types):gst_structure_get_array()gst_structure_set_array()gst_structure_get_list()gst_structure_set_list()
a new 'dynamic type' registry factory type was added to register dynamically loadable GType types. This is useful for automatically loading enum/flags types that are used in caps, such as for example the GstVideoMultiviewFlagsSet type used in multiview video caps.
there is a new GstProxyControlBinding for use with GstController. This allows proxying the control interface from one property on one GstObject to another property (of the same type) in another GstObject. So e.g. in parent-child relationship, one may need to call gst_object_sync_values() on the child and have a binding (set elsewhere) on the parent update the value. This is used in glvideomixer and glsinkbin for example, where sync_values() on the child pad or element will call sync_values() on the exposed bin pad or element.
Note that this doesn't solve GObject property forwarding, that must be taken care of by the implementation manually or using GBinding.
gst_base_parse_drain() has been made public for subclasses to use.
`gst_base_sink_set_drop_out_of_segment()' can be used by subclasses to prevent GstBaseSink from dropping buffers that fall outside of the segment.
gst_calculate_linear_regression() is a new utility function to calculate a linear regression.
gst_debug_get_stack_trace is an easy way to retrieve a stack trace, which can be useful in tracer plugins.
allocators: the dmabuf allocator is now sub-classable, and there is a new GST_CAPS_FEATURE_MEMORY_DMABUF define.
video decoder subclasses can use the newly-added function gst_video_decoder_allocate_output_frame_with_params() to pass a GstBufferPoolAcquireParams to the buffer pool for each buffer allocation.
the video time code API has gained a dedicated GstVideoTimeCodeInterval type plus related API, including functions to add intervals to timecodes.
There is a new libgstbadallocators-1.0 library in gst-plugins-bad, which may go away again in future releases once the GstPhysMemoryAllocator interface API has been validated by more users and was moved to libgstallocators-1.0 from gst-plugins-base.
GstPlayer:
New API has been added to:
get the number of audio/video/subtitle streams:gst_player_media_info_get_number_of_streams()gst_player_media_info_get_number_of_video_streams()gst_player_media_info_get_number_of_audio_streams()gst_player_media_info_get_number_of_subtitle_streams()
enable accurate seeking: gst_player_config_set_seek_accurate() and gst_player_config_get_seek_accurate()
get a snapshot image of the video in RGBx, BGRx, JPEG, PNG or native format: gst_player_get_video_snapshot()
selecting use of a specific video sink element (gst_player_video_overlay_video_renderer_new_with_sink())
If the environment variable GST_PLAYER_USE_PLAYBIN3 is set, GstPlayer will use the still-experimental playbin3 element and the GstStreams API for playback.
Miscellaneous changes:
video caps for interlaced video may contain an optional "field-order" field now in the case of interlaced-mode=interleaved to signal that the field order is always the same throughout the stream. This is useful to signal to muxers such as mp4mux. The new field is parsed from/to GstVideoInfo of course.
video decoder and video encoder base classes try harder to proxy interlacing, colorimetry and chroma-site related fields in caps properly.
The buffer stored in the PROTECTION events is now left unchanged. This is a change of behaviour since 1.8, especially for the mssdemux element which used to decode the base64 parsed data wrapped in the protection events emitted by the demuxer.
PROTECTION events can now be injected into the pipeline from the application; source elements deriving from GstBaseSrc will forward those downstream now.
The DASH demuxer is now correctly parsing the MSPR-2.0 ContentProtection nodes and emits Protection events accordingly. Applications relying on those events might need to decode the base64 data stored in the event buffer before using it.
The registry can now also be disabled by setting the environment variable GST_REGISTRY_DISABLE=yes, with similar effect as the GST_DISABLE_REGISTRY compile time switch.
Seeking performance with gstreamer-vaapi based decoders was improved. It would recreate the decoder and surfaces on every seek which can be quite slow.
more robust handling of input caps changes in videoaggregator-based elements such as compositor.
Lots of adaptive streaming-related fixes across the board (DASH, MSS, HLS). Also:
mssdemux, the Microsoft Smooth Streaming demuxer, has seen various fixes for live streams, duration reporting and seeking.
The DASH manifest parser now extracts MS PlayReady ContentProtection objects from manifests and sends them downstream as PROTECTION events. It also supports multiple Period elements in external xml now.
gst-libav was updated to ffmpeg 3.3 but should still work with any 3.x version.
GstEncodingProfile has been generally enhanced so it can, for example, be used to get possible profiles for a given file extension. It is now possible to define profiles based on element factory names or using a path to a .gep file containing a serialized profile.
audioconvert can now do endianness conversion in-place. All other conversions still require a copy, but e.g. sign conversion and a few others could also be implemented in-place now.
The new, experimental playbin3 and urisourcebin elements got many bugfixes and improvements and should generally be closer to a full replacement of the old elements.
interleave now supports > 64 channels.
OpenCV elements, grabcut and retinex has been ported to use GstOpencvVideoFilter base class, increasing code reuse and fixing buffer map/unmap issues. Redundant copie of images has been removed in edgedetect, cvlaplace and cvsobel. This comes with various cleanup and Meson support.
OpenGL integration:
As usual the GStreamer OpenGL integration library has seen numerous fixes and performance improvements all over the place, and is hopefully ready now to become API stable and be moved to gst-plugins-base during the 1.14 release cycle.
The GStreamer OpenGL integration layer has also gained support for the Vivante EGL FB windowing system, which improves performance on platforms such as Freescale iMX.6 for those who are stuck with the proprietary driver. The qmlglsink element also supports this now if Qt is used with eglfs or wayland backend, and it works in conjunction with gstreamer-imx of course.
various qmlglsrc improvements
Tracing framework and debugging improvements:
New tracing hooks have been added to track GstMiniObject and GstObject ref/unref operations.
The memory leaks tracer can optionally use this to retrieve stack traces if enabled with e.g. GST_TRACERS=leaks(filters="GstEvent,GstMessage",stack-traces-flags=full)
The GST_DEBUG_FILE environment variable, which can be used to write the debug log output to a file instead of printing it to stderr, can now contain a name pattern, which is useful for automated testing and continuous integration systems. The following format specifiers are supported:%p: will be replaced with the PID%r: will be replaced with a random number, which is useful for instance when running two processes with the same PID but in different containers.
Tools:
gst-inspect-1.0 can now list elements by type with the new --types command-line option, e.g. gst-inspect-1.0 --types=Audio/Encoder will show a list of audio encoders.
gst-launch-1.0 and gst_parse_launch() have gained a new operator (:) that allows linking all pads between two elements. This is useful in cases where the exact number of pads or type of pads is not known beforehand, such as in the uridecodebin : encodebin scenario, for example. In this case, multiple links will be created if the encodebin has multiple profiles compatible with the output of uridecodebin.
gst-device-monitor-1.0 now shows a gst-launch-1.0 snippet for each device that shows how to make use of it in a gst-launch-1.0 pipeline string.
GStreamer RTSP server:
The RTSP server now also supports Digest authentication in addition to Basic authentication.
The GstRTSPClient class has gained a pre-*-request signal and virtual method for each client request type, emitted in the beginning of each rtsp request. These signals or virtual methods let the application validate the requests, configure the media/stream in a certain way and also generate error status codes in case of an error or a bad request.
GStreamer VAAPI:
GstVaapiDisplay now inherits from GstObject, thus the VA display logging messages are better and tracing the context sharing is more readable.
When uploading raw images into a VA surfaces now VADeriveImages are tried fist, improving the upload performance, if it is possible.
The decoders and the post-processor now can push dmabuf-based buffers to downstream under certain conditions. For example:
GST_GL_PLATFORM=egl gst-play-1.0 video-sample.mkv --videosink=glimagesink
Refactored the wrapping of VA surface into gstreamer memory, adding lock when mapping and unmapping, and many other fixes.
Now vaapidecodebin loads vaapipostproc dynamically. It is possible to avoid it usage with the environment variable GST_VAAPI_DISABLE_VPP=1.
Regarding encoders: they have primary rank again, since they can discover, in run-time, the color formats they can use for upstream raw buffers and caps renegotiation is now possible. Also the encoders push encoding info downstream via tags.
About specific encoders: added constant bit-rate encoding mode for VP8 and H265 encoder handles P010_10LE color format.
Regarding decoders, flush operation has been improved, now the internal VA encoder is not recreated at each flush. Also there are several improvements in the handling of H264 and H265 streams.
VAAPI plugins try to create their on GstGL context (when available) if they cannot find it in the pipeline, to figure out what type of VA Display they should create.
Regarding vaapisink for X11, if the backend reports that it is unable to render correctly the current color format, an internal VA post-processor, is instantiated (if available) and converts the color format.
GStreamer Editing Services and NLE
Enhanced auto transition behaviour
Fix some races in nlecomposition
Allow building with msvc:
Added a UNIX manpage for ges-launch
API changes:Added ges_deinit (allowing the leak tracer to work properly)Added ges_layer_get_clips_in_intervalFinally hide internal symbols that should never have been exposed
GStreamer validate:
Port gst-validate-launcher to python 3
gst-validate-launcher now checks if blacklisted bugs have been fixed on bugzilla and errors out if it is the case
Allow building with msvc
Add ability for the launcher to run GStreamer unit tests
Added a way to activate the leaks tracer on our tests and fix leaks
Make the http server multithreaded
New testsuite for running various test scenarios on the DASH-IF test vectors
GStreamer Python Bindings:
Overrides has been added for IntRange, Int64Range, DoubleRange, FractionRange, Array and List. This finally enables Python programmers to fully read and write GstCaps objects.
Build and Dependencies:
Meson build files are now disted in tarballs, for jhbuild and so distro packagers can start using it. Note that the Meson-based build system is not 100% feature-equivalent with the autotools-based one yet.
Some plugin filenames have been changed to match the plugin names: for example the file name of the encoding plugin in gst-plugins-base containing the encodebin element was libgstencodebin.so and has been changed to libgstencodebin.so. This affects only a handful of plugins across modules.
Developers who install GStreamer from source and just do make install after updating the source code, without doing make uninstall first, will have to manually remove the old installed plugin files from the installation prefix, or they will get 'Cannot register existing type' critical warnings.
Most of the docbook-based documentation (FAQ, Application Development Manual, Plugin Writer's Guide, design documents) has been converted to markdown and moved into a new gst-docs module. The gtk-doc library API references and the plugins documentation are still built as part of the source modules though.
GStreamer core now optionally uses libunwind and libdw to generate backtraces. This is useful for tracer plugins used during debugging and development.
There is a new libgstbadallocators-1.0 library in gst-plugins-bad (which may go away again in future releases once the GstPhysMemoryAllocator interface API has been validated by more users).
gst-omx and gstreamer-vaapi modules can now also be built using the Meson build system.
The qtkitvideosrc element for macOS was removed. The API is deprecated since 10.9 and it wasn't shipped in the binaries since a few releases.
Platform-specific improvements:
Android
androidmedia: add support for VP9 video decoding/encoding and Opus audio decoding (where supported)
OS/X and iOS:
avfvideosrc, which represents an iPhone camera or, on a Mac, a screencapture session, so far allowed you to select an input device by device index only. New API adds the ability to select the position (front or back facing) and device-type (wide angle, telephoto, etc.). Furthermore, you can now also specify the orientation (portrait, landscape, etc.) of the videostream.
Windows:
dx9screencapsrc can now optionally also capture the cursor.

New in GStreamer 1.10.0 (Nov 1, 2016)

New in GStreamer 1.8.1 (Apr 20, 2016)

New in GStreamer 1.8.0 (Mar 24, 2016)

New in GStreamer 1.6.3 (Jan 20, 2016)

New in GStreamer 1.6.2 (Dec 14, 2015)

New in GStreamer 1.6.1 (Oct 31, 2015)

New in GStreamer 1.6.0 (Sep 26, 2015)

Stereoscopic 3D and multiview video support
Trick mode API for key-frame only fast-forward/fast-reverse playback etc.
Improved DTS (decoding timestamp) vs. PTS (presentation timestamp) handling to account for negative DTS
New GstVideoConverter API for more optimised and more correct conversion of raw video frames between all supported formats, with rescaling
v4l2src now supports renegotiation
v4l2transform can now do scaling
V4L2 Element now report Colorimetry properly
Easier chunked recording of MP4, Matroska, Ogg, MPEG-TS: new splitmuxsink and multifilesink improvements
Content Protection signalling API and Common Encryption (CENC) support for DASH/MP4
Many adaptive streaming (DASH, HLS and MSS) improvements
New PTP and NTP network client clocks and better remote clock tracking stability
High-quality text subtitle overlay at display resolutions with glimagesink or gtkglsink
RECORD support for the GStreamer RTSP Server
Retransmissions (RTX) support in RTSP server and client
RTSP seeking support in client and server has been fixed
RTCP scheduling improvements and reduced size RTCP support
MP4/MOV muxer acquired a new "robust" mode of operation which attempts to keep the output file in a valid state at all times
Live mixing support in aggregator, audiomixer and compositor was improved a lot
compositor now also supports rescaling of inputs streams on the fly
New audiointerleave element with proper input synchronisation and live input support
Blackmagic Design DeckLink capture and playback card support was rewritten from scratch; 2k/4k support; mode sensing
KLV metadata support in RTP and MPEG-TS
H.265 video encoder (x265), decoders (libav, libde265) and RTP payloader and depayloaders
New DTLS plugin and SRTP/DTLS support
OpenGL3 support, multiple contexts and context propagation, 3D video, transfer/conversion separation, subtitle blending
New OpenGL-based QML video sink, Gtk GL video sink, CoreAnimation CAOpenGLLayerSink video sink
gst-libav switched to ffmpeg as libav-provider, gains support for 3D/multiview video, trick modes, and the CAVS codec
GstHarness API for unit tests
gst-editing-services got a completely new ges-launch-1.0 interface, improved mixing support and integration into gst-validate
gnonlin has been deprecated in favor of nle (Non Linear Engine) in gst-editing-services
gst-validate has a new plugin system, an extensive default testsuite, support for concurrent test runs and valgrind support
cerbero build tool for SDK binary packages gains new 'bundle-source' command
Various improvements to the Android, iOS, OS X and Windows platform support

New in GStreamer 1.5.2 (Jun 26, 2015)

New in GStreamer 1.4.5 (Dec 21, 2014)

GStreamer core:
736969 : queue2: dead lock when buffering
738092 : basesink: clamp reported position based on direction
740001 : task: race condition when pausing and stopping
GStreamer Plugins Base:
741420 : video pools: should update size in configuration after applying alignment
715050 : add typefinder for audio/x-audible
739544 : tcp: Add test and fix memory leak in tcp elements
739840 : typefind should recognize Apple Core Audio Format (CAF)
740556 : videodecoder: don't complain when DTS != PTS on keyframes
740675 : playsink: continues playback, reset mute property
740730 : rtspconnection: don't remove child source if parent source is already destroyed
740853 : audiodecoder: Push pending events before sending EOS.
740952 : alsa: NetBSD fixes
741045 : audiorate can can lose timestamp precision in some cases
741198 : playbin: leaks GstPads
GStreamer Plugins Good:
711437 : apev2mux: should not put APEv2 tags at beginning of WavPack files
726194 : v4l2src does not cope well when a capture card is sometimes interlaced, sometimes progressive at the same resolution
736397 : flvdemux: Per-stream resyncing behavior causes A/V-desyncs
737603 : v4l2bufferpool: set buffer interlace flags when field is V4L2_FIELD_INTERLACED
739476 : vpx: fails to build against libvpx from git
739722 : matroskamux: Thread safe register GstMatroskamuxPad
739789 : v4l2allocator: fix error message if allocator is already active
739791 : v4l2bufferpool: unref pool when v4l2_allocator_new() fails
739792 : v4l2allocator: fix gst_v4l2_allocator_stop prototype
739996 : videomixer: Drops a lot of frames, if one of the sources is live
740040 : v4l2src: Fails in presence of V4L2_BUF_FLAG_ERROR
740392 : rtspsrc: mikey base64 decoded key-mgmt leak
740407 : qtmux limits capture to 4096x4096
740633 : v4l2src: RW io-mode is broken
740636 : v4l2src: framerate is not always set on driver
740671 : aspectratiocrop: crop needs to be reset when video size changes
740905 : v4l2: still has 1 include to linux/videodev.h
741271 : rtph264pay: Buffer leak in H.264 payloader when using SPS/PPS
741381 : rtph264pay: Race condition may cause crash when going from PAUSED- > READY
741407 : deinterlace: in query_caps return only supported formats if filter is interlaced
737579 : v4l2object: set colorspace for output devices
739754 : v4l2bufferpool: Should validate that all memories are writeable before queueing back
GStreamer Plugins Bad:
722764 : rawparse: fix SEEKING query handling
729768 : mpegtsbase: Raise limit to read more bytes initially to find PCR
739152 : gl/cocoa: build with GNUStep fails
740191 : dvbbasesink: segfaults on 32-bit (rpi)
740201 : gl/cocoa: Does not compile on OSX < 10.7 anymore
740451 : srtpdec: leaks rtp/rtcp sink events
740953 : configure.ac: unportable test(1) comparison operator
741321 : opusparse: fix header parsing esp. of encoded output of libopus
GStreamer RTSP Server:
739481 : rtsp-stream: leaks srtp decoder when leaving rtpbin

New in GStreamer 1.4.4 (Nov 10, 2014)

New in GStreamer 1.4.1 (Aug 28, 2014)

New in GStreamer 1.4 RC2 (Jul 12, 2014)

New in GStreamer 1.4 RC1 (Jun 30, 2014)

New API:
GstMessageType has GST_MESSAGE_EXTENDED added. All types before that can be used together as a flags type as before, but from that message onwards the types are just counted incrementally. This was necessary to be able to add more message types. In 2.0 GstMessageType will just become an enum and not a flags type anymore.
GstDeviceMonitor for device probing, e.g. to list all available audio or video capture devices. This is the replacement for GstPropertyProbe from 0.10.
Events accumulate the running-time offset now when travelling through pads, as set by the gst_pad_set_offset() function. This allows to compensate for this in the QOS event for example.
GstBuffer has a new flag "tag-memory" that is set automatically when memory is added or removed to a buffer. This allows buffer pools to detect if they can recycle a buffer or need to reset it first.
GstToc has new API to mark GstTocEntries as loops.
A not-authorized resource error has been defined to notify applications that accessing the resource has failed because of missing authorization and to distinguish this case from others. This change is actually already in 1.2.4.
GstPad has a new flag "accept-intersect", that will let the default ACCEPT_CAPS query handler do an intersection instead of subset check. This is interesting for parser elements that can handle incomplete caps.
GstCollectPads has support for flushing and a default handler for SEEK events now.
New GstFlowAggregator helper object that simplifies handling of flow returns in elements with multiple source pads. Additionally GstPad now always stores the last flow return and provides an API to retrieve it.
GstSegment has new API to offset the running time by a specific value and this is used in GstPad to allow positive and negative offsets in gst_pad_set_offset() in all situations.
Support for h265/HEVC and VP8 has been added to the codec utils and codec parsers library, and was integrated into various elements.
API for adjusting the TLS validation of RTSP connection has been added.
The RTSP and SDP library has MIKEY (RFC 3830) support now, and there is API to distinguish between the different RTSP profiles.
API to access RTP time information and statistics.
Support for auxiliary streams was added to rtpbin.
Support for tiled, raw video formats has been added.
GstVideoDecoder and GstAudioDecoder have API to help aggregating tag events and merge custom tags into them consistently.
GstBufferPool has support for flushing now.
playbin/playsink has support for application provided audio and video filters.
GstDiscoverer has new and simplified API to get details about missing plugins and information to pass to the plugin installer.
The GL library was merged from gst-plugins-gl to gst-plugins-bad, providing a generic infrastructure for handling GL inside GStreamer pipelines and a plugin with some elements using these, especially a video sink. Supported platforms currently are Android, Cocoa (OS X), DispManX (Raspberry Pi), EAGL (iOS), WGL (Windows) and generic X11, Wayland and EGL platforms. This replaces eglglessink and also is supposed to replace osxvideosink.
New GstAggregator base class in gst-plugins-bad. This is supposed to replace GstCollectPads in the future and fix long-known shortcomings in its API. Together with the base class some elements are provided already, like a videomixer (compositor).
Major changes:
New plugins and elements:
v4l2videodec element for accessing hardware codecs on platforms that make them accessible via V4L2, e.g. Samsung Exynos. This comes together with major refactoring of the existing V4L2 elements and the corresponding infrastructure. The v4l2videodec element replaces the mfcdec element.
New downloadbuffer element that replaces the download buffering feature of queue2. Compared to queue2's code it is much simpler and only for this single use case. A noteworthy new feature is that it's downloading gaps in the already downloaded stream parts when nothing else is to be downloaded. This is now used by playbin when download buffering is enabled.
rtpstreampay and rtpstreamdepay elements for transmitting RTP packets over a stream API (e.g. TCP) according to RFC 4571.
rtprtx elements for standard compliant implementation of retransmissions, integrated into the rtpmanager plugin.
audiomixer element that mixes multiple audio streams together into a single one while keeping synchronization. This is planned to become the replacement of the adder element.
OpenNI2 plugin for 3D cameras like the Kinect camera.
OpenEXR plugin for decoding high-dynamic-range EXR images.
curlsshsink and curlsftpsink to write files via SSH/SFTP.
videosignal, ivfparse and sndfile plugins ported from 0.10.
avfvideosrc, vtdec and other elements were ported from 0.10 and are available on OS X and iOS now.
Other changes:
gst-libav now uses libav 10.1, and gained support for H265/HEVC.
Support for hardware codecs and special memory types has been improved with bugfixes and feature additions in various plugins and base classes.
Various bugfixes and improvements to buffering in queue2 and multiqueue elements.
dvbsrc supports more delivery mechanisms and other features now, including DVB S2 and T2 support.
The MPEGTS library has support for many more descriptors.
Major improvements to tsdemux and tsparse, especially time and seeking related.
souphttpsrc now has support for keep-alive connections, compression, configurable number of retries and configuration for SSL certificate validation.
hlsdemux has undergone major refactoring and works more reliable now and supports more HLS features like trick modes. Also fragments are pushed downstream while they're downloaded now instead of waiting for each fragment to finish.
dashdemux and mssdemux are now also pushing fragments downstream while they're downloaded instead of waiting for each fragment to finish.
videoflip can automatically flip based on the orientation tag.
openjpeg supports the OpenJPEG2 API.
waylandsink was refactored and should be more useful now. It also includes a small library which most likely is going to be removed in the future and will result in extensions to the GstVideoOverlay interface.
gst-rtsp-server supports SRTP and MIKEY now.
gst-libav encoders are now negotiating any profile/level settings with downstream via caps.
Lots of fixes for coverity warnings all over the place.
Negotiation related performance improvements.
800+ fixed bug reports, and many other bug fixes and other improvements everywhere that had no bug report.
Things to look out for:
The eglglessink element was removed and replaced by the glimagesink element.
The mfcdec element was removed and replaced by v4l2videodec.
osxvideosink is only available in OS X 10.6 or newer.
On Android the namespace of the automatically generated Java class for initialization of GStreamer has changed from com.gstreamer to org.freedesktop.gstreamer to prevent namespace pollution.
On iOS you have to update your gst_ios_init.h and gst_ios_init.m in your projects from the one included in the binaries if you used the GnuTLS GIO module before. The loading mechanism has slightly changed.

New in GStreamer 1.2.4 (Apr 22, 2014)

GStreamer core:
724373 : Queue2 truncates its temp file when pipeline is paused
725517 : docs: Fix typos and remove unknown annotations
725809 : ghostpad: rare crash because of missing reference count on its target pad
727253 : parse: Bison generated file included in the release tarballs causes compile errors
727883 : baseparse: Memory leak of queue frames
GStreamer Plugins Base:
693263 : typefinding: MPEG-2 video ES detected as H.263
683504 : playsink: deadlock when disabling subtitles and suboptimal disabling of subtitles
700770 : typefinding: mp3 file mis-detected as h263 video
723597 : tagdemux: Seek event in GST_FORMAT_TIME are converted to BYTES to early
724633 : oggdemux: ignores last page in push mode
724720 : rtspconnection: not possible to disconnect/reconnect read connection in tunneled mode
725313 : rtspconnection: closed() callback is never called in tunneled mode
725644 : typefinding: mp3 file is misdetected as H.263
726642 : rtspconnection: minor memory leak in error handling
727025 : adder: rework the logic to check if eos has to be sent.
GStreamer Plugins Good:
725104 : qtdemux: reverse playback and video stream switching failure
722185 : souphttpsrc: racy " server does not support seeking " error
724619 : crash when reading the device name property of pulsesink
725124 : rtspsrc: Fix deadlock when task creation is no successful
725712 : rtpsession: Crash when RTCP FIR received with unknown SSRC
725860 : v4l2src: Fix using v4l2src with Hauppauge HDPVR video capture device
726777 : rtpjpegpay: payload size not correctly calculated
728017 : [regression]eos event could not be send out from gstrtpjitterbuffer.
728041 : rtph264depay: marks all output buffers as delta units when outputting avc format
724638 : aacparse : Missing resilience when no audio frame is found
727329 : check: souphttpsrc: unknown type name ‘SoupStatus’
GStreamer Plugins Bad:
724013 : Don't hardcode /usr/share/sounds/sf2 path in fluiddec
725137 : hlsdemux: fails to compute media playlist URL if there is a query parameter
725140 : hlsdemux: fails to correctly parse CODECS and RESOLUTION
GStreamer libav Plugins:
727779 : avdec_h264, matroskademux: crash while seeking (1.2 regression)

New in GStreamer 1.2.2 (Dec 28, 2013)

New in GStreamer 1.2.0 (Sep 25, 2013)

New API:
GstContext negotiation / sharing / announcing for sharing a generic context between elements, e.g. a display handle
GL texture upload conversion meta for allowing different buffer types to be converted to an OpenGL texture
GstCapsFeatures as extension to GstCaps for allowing the negotiation of specific memory or meta requirements between elements
GstMemory flags for contiguous and non-mappable memory
The stream-start event has optional flags now, e.g. for signalling sparse streams
The stream-start even has an optional group-id field now to signal all streams that should be played together
Allocators library in gst-plugins-base, currently only with generic dmabuf memory support
insertbin library for easier handling of dynamically linked pipelines (in -bad for now)
EGL helper library (in -bad for now)
MPEG-TS data structure library (in -bad for now)
New GstVideoRegionOfInterestMeta to describe a region of interest on video frames.
GstVideoDecoder/Encoder has new ::flush() vfunc to replace the ill-defined ::reset() vfunc.
The URI query allows to query the redirected URI now.
Major changes:
New tool: gst-play-1.0 in gst-plugins-base for basic playback testing on the command line.
New plugins:
mssdemux for Microsoft Smooth Streaming
dashdemux for DASH adaptive streaming protocol
bluez for interaction with Bluetooth devices
openjpeg for JPEG2000 decoding and encoding
daala for experimental Daala decoding and encoding
vpx plugin has experimental VP9 decoding and encoding support
webp plugin for WebP decoding (encoding to be added later)
Various others: yadif, srtp, sbc, fluidsynth, midiparse, mfc, ivtv, accuraterip and audiofxbad
Moved plugins:
dtmf, vp8rtp, scaletempo and rtpmux plugins are in gst-plugins-good now
Video:
Fix handling of interlaced video in converters such as videoscale and videoconvert (e.g. scale both fields independently)
videoconvert will try harder to minimise quality losses when conversion is necessary
The experimental GstSurfaceConverter, GstSurfaceMeta and GstVideoContext APIs from the (confusingly-named) libgstbasevideo-1.0 library in gst-plugins-bad have now been removed and been replaced by new APIs in GStreamer Core and gst-plugins-base (see above). Since that was all that was left in this library, the entire experimental libgstbasevideo-1.0 library has been removed from gst-plugins-bad
Chroma subsampling and chroma siting conversion is better handled in videoconvert and the support for interlaced video was improved.
New pinwheel and spoke patterns in videotestsrc
videomixer can now accept different video formats on its sinkpads and converts to a common format during mixing
Audio:
audioconvert will try harder to minimise quality losses when conversion is necessary
adder now allows muting/unmuting of its input streams, and also per-input stream volume
pulseaudio elements can switch between devices during playback now
aacparse can convert between ADTS←→RAW
Platform specific changes:
Caps, events, etc. are now printed in the GStreamer debug logs with their content instead of just the pointer address even on non-glibc platforms (e.g. Windows, OSX, Android).
Network elements (UDP/TCP) now work better with platforms, where IPv6 sockets can't handle IPv4 (e.g. Windows)
Linux/BSD: v4l2 had many improvements and cleanups
Other changes:
gst-libav now uses libav 9
Static linking of plugins is supported now (also in 1.0.7)
rtspsrc: add support for NetClientClock: when the server suggests a GstNetTimeProvider in the SDP, set up a GstNetClientClock that slaves to the remote clock and suggest this clock in provide_clock. Simplifies synchronized playback of a resource from an RTSP server. gst-rtsp-server now supports adding this to the SDP and can provide a network clock
RTP retransmission / NACK support and big RTP jitterbuffer improvements
SRTP and DTLS support
Changes to many elements and core to use the correct sticky event order and also not lose any important sticky events during flushing
>1000 fixed bug reports, and many other bug fixes and other improvements everywhere that had no bug report
Things to look out for:
Single header includes for all libraries, e.g. #include - this was needed for some bindings.
Stricter (correct) caps subset checking in some cases where this was not correct before. Caps will now always fail to be a compatible subset of another set of caps if the subset caps are missing some fields that the superset caps have. This might lead to not-negotiated errors if caps are incomplete now. However, it also prevents possible data corruption caused by piping data formatted in an incompatible/unexpected way into some elements. Check your h264 caps for stream-format and alignment fields and AAC caps for the stream-format field. This change will also be included in the next stable 1.0.8 release.
Stricter checking for missing events and correct sticky event order (stream-start, caps, segment) in some places; this is not enabled in stable releases by default, but you may get warnings when using git builds, development releases or when compiling with -UG_DISABLE_ASSERT in CFLAGS
x264enc now outputs data in byte-stream by default if downstream has ANY caps (e.g. appsink without caps set, filesink, udpsink, tcpserversink etc.)
The MPEG TS demuxer posts messages contain the PMT, PAT, etc. in a different format now. This new format uses the data structures from the new MPEGTS library
The GstContext API has changed between 1.1.4 and 1.1.90

New in GStreamer 1.1.4 (Aug 29, 2013)

New API:
GstContext negotiation / sharing / announcing for sharing a generic context between elements, e.g. a display handle
GL texture upload conversion meta for allowing different buffer types to be converted to an OpenGL texture
GstCapsFeatures as extension to GstCaps for allowing the negotiation of specific memory or meta requirements between elements
GstMemory flags for contiguous and non-mappable memory
The stream-start event has optional flags now, e.g. for signalling sparse streams
The stream-start even has an optional group-id field now to signal all streams that should be played together
Allocators library in gst-plugins-base, currently only with generic dmabuf memory support
insertbin library for easier handling of dynamically linked pipelines (in -bad for now)
EGL helper library (in -bad for now)
MPEG-TS data structure library (in -bad for now)
New GstVideoRegionOfInterestMeta to describe a region of interest on video frames.
GstVideoDecoder/Encoder has new ::flush() vfunc to replace the ill-defined ::reset() vfunc.
The URI query allows to query the redirected URI now.
Major changes:
New tool: gst-play-1.0 in gst-plugins-base for basic playback testing on the command line.
New plugins:
mssdemux for Microsoft Smooth Streaming
dashdemux for DASH adaptive streaming protocol
bluez for interaction with Bluetooth devices
openjpeg for JPEG2000 decoding and encoding
daala for experimental Daala decoding and encoding
vpx plugin has experimental V9 decoding and encoding support
webp plugin for WebP decoding (encoding to be added later)
Various others: yadif, srtp, sbc, fluidsynth, midiparse, mfc, ivtv, accuraterip and audiofxbad
Moved plugins:
dtmf, vp8rtp, scaletempo and rtpmux plugins are in gst-plugins-good now
Video:
Fix handling of interlaced video in converters such as videoscale and videoconvert (e.g. scale both fields independently)
videoconvert will try harder to minimise quality losses when conversion is necessary
The experimental GstSurfaceConverter, GstSurfaceMeta and GstVideoContext APIs from the (confusingly-named) libgstbasevideo-1.0 library in gst-plugins-bad have now been removed and been replaced by new APIs in GStreamer Core and gst-plugins-base (see above). Since that was all that was left in this library, the entire experimental libgstbasevideo-1.0 library has been removed from gst-plugins-bad.
Chroma subsampling and siting conversion is better handled in videoconvert
New pinwheel and spoke patterns in videotestsrc
Audio:
adder now allows muting/unmuting of its input streams, and also per-input stream volume
pulseaudio elements can switch between devices during playback now
aacparse can convert between ADTS←→RAW
Platform specific changes:
Caps, events, etc. are now printed in the GStreamer debug logs with their content instead of just the pointer address even on non-glibc platforms (e.g. Windows, OSX, Android).
Network elements (UDP/TCP) now work better with platforms, where IPv6 sockets can't handle IPv4 (e.g. Windows)
Windows: d3dvideosink provides a bufferpool to upstream elements
Linux/BSD: v4l2 had many improvements and cleanups
Other changes:
gst-libav now uses libav 9
Static linking of plugins is supported now (also in 1.0.7)
rtspsrc: add support for NetClientClock: when the server suggests a GstNetTimeProvider in the SDP, set up a GstNetClientClock that slaves to the remote clock and suggest this clock in provide_clock. Simplifies synchronized playback of a resource from an RTSP server. gst-rtsp-server now supports adding this to the SDP and can provide a network clock
RTP retransmission / NACK support and big RTP jitterbuffer improvements
SRTP and DTLS support
Changes to many elements and core to use the correct sticky event order and also not lose any important sticky events during flushing
>1000 fixed bug reports, and many other bug fixes and other improvements everywhere that had no bug report
Things to look out for:
Single header includes for all libraries, e.g. #include - this was needed for some bindings.
Stricter (correct) caps subset checking in some cases where this was not correct before. Caps will now always fail to be a compatible subset of another set of caps if the subset caps are missing some fields that the superset caps have. This might lead to not-negotiated errors if caps are incomplete now. However, it also prevents possible data corruption caused by piping data formatted in an incompatible/unexpected way into some elements. Check your h264 caps for stream-format and alignment fields and AAC caps for the stream-format field. This change will also be included in the next stable 1.0.8 release.
Stricter checking for missing events and correct sticky event order (stream-start, caps, segment) in some places; this is not enabled in stable releases by default, but you may get warnings when using git builds, development releases or when compiling with -UG_DISABLE_ASSERT in CFLAGS
x264enc now outputs data in byte-stream by default if downstream has ANY caps (e.g. appsink without caps set, filesink, udpsink, tcpserversink etc.)

New in GStreamer 1.0.8 (Jul 12, 2013)

GStreamer core changes since 1.0.7:
basesink: improve position reporting without clock
fix caps leak in typefind/decodebin/playbin
gobject-introspection fixes for bindings
GStreamer core bugs fixed since 1.0.7:
693365 : gst_structure_is_subset false positive
702617 : buffer: Wrong size/index handling when merging memory
702778 : REGRESSION : Backward seeking doesn't work with mp3 files.
703562 : Missing few allow-none annotation
GStreamer Plugins Base changes since 1.0.7:
tag: ignore malformed ID3v2 TDAT frames
GStreamer Plugins Base bugs fixed since 1.0.7:
636245 : riff: for ADPCM codecs the average bitrate should be calculated instead of trusting the format header
688803 : playbin: converters don't work? not-negotiated error with non-1/1 PAR and ximagesink
690420 : decodebin: Race between GstBin and decodebin trying to change states of child elements
698896 : liveadder: What is the unit for " latency " ?
699923 : typefind: fix detection of HLS playlists with alternative renditions
701976 : flvdemux: Forwards CAPS events from upstream
703128 : videotestsrc leaks memory
703283 : id3: gst_date_time_new: assertion `(month > 0 & & month < = 12) || month == -1' failed with malformed TDAT frames
699794 : uridecodebin: Leaking queue2 elements in playbin gapless mode
GStreamer Plugins Good changes since 1.0.7:
pngenc: fix massive memory leak
pngdec: parse incoming data into frames before decoding
osvideo: many osxvideosink fixes
udpsink, multiudpsink, dynudpsink: bind socket before using it, fixes sending udp on windows
GStreamer Plugins Good bug fixes since 1.0.7:
682110 : qtdemux: discont flag set on multiple buffers in push mode
692400 : udpsrc: fix socket options not getting set on windows, resulting in packet drop in high bitrate movie
693727 : rtpvrawpay/depay negotiation broken
699260 : pngenc: unmap source frame when done
699303 : matroskademux: stream-format=raw missing from aac caps
699314 : rtph264pay: CRITICAL **: gst_adapter_map: assertion `size > 0' failed
700047 : pngdec: make decoding work without png parser
700382 : qtdemux: handle 96kHz/24 bits ALAC audio
700514 : rtpmp4apay: clear config buffer before using it
700878 : udpsink: Not sending anything on Windows
701586 : rtspsrc memleak
702167 : matroskademux: missing mutex unlock leading to wrong STREAM_LOCK count
702457 : rtpmp2tdepay: unable to depay MPEG2-TS RTP streams from GStreamer 0.10
702705 : rtspsrc does not pause properly (race condition)
702732 : v4l2: Does not link statically on linux
703076 : flvdemux: Add flvversion 1 to flash-video caps
703100 : osxvideosink: deadlock on re-use
703171 : rtph264pay: segfault because of double buffer unmap on error
703729 : osxvideosink doesn't display video anymore if once set to READY
691419 : osxvideosink: doesn't close internal window in case of window-id assignment
GStreamer Plugins Ugly changes since 1.0.7:
lamemp3enc: fix timestamping of outgoing buffers if the encoder resamples internally, which fixes transcoding pipelines deadlocking after a while
GStreamer Plugins Bad changes since 1.0.7:
rfbsrc, neonhttpsrc, ofa, and openal plugins ported to 1.0
mpegvideoparse: don't announce incomplete source caps
tsdemux: many fixes
GStreamer Plugins Bad bugs fixed since 1.0.7:
702495 : sdpdemux fails if not explicitly added to the pipeline
674536 : tsdemux: Freeze on pts-wrap with streaming sources
685103 : mpegvideoparse: wrong pixel-aspect-ratio
695412 : mpegtsmux AAC ADTS header seems incorrect
695879 : mpegvideoparse: outputs incomplete caps with different mpegversion before outputting proper caps
697283 : mpegdemux: accept ID_PRIVATE_STREAM_1 to avoid loosing sync
699786 : mpegtsmux: memory leak when using prepare_func
700038 : rfbsrc: port to 1.0
700402 : openalsink: 'AL_FORMAT_MONO_ALAW_EXT' undeclared
702597 : shmsink: events not propagated to basesink
GStreamer Libav Plugins changes since 1.0.7:
gst-libav: internal libav snapshot version bumped to v0.8.8

New in GStreamer 1.0.6 (Mar 23, 2013)

New in GStreamer 1.0.5 (Jan 8, 2013)

New in GStreamer 1.0.4 (Dec 19, 2012)

New in GStreamer 1.0.3 (Nov 21, 2012)

New in GStreamer 1.0.2 (Oct 25, 2012)

New in GStreamer 1.0.1 (Oct 8, 2012)

New in GStreamer 1.0.0 (Sep 25, 2012)

New in GStreamer 0.10.31 (Dec 2, 2010)

bin: add "message-forward" property to force forwarding of messages that would usually be filtered such as ASYNC_DONE or EOS
bin: improve tracking of source elements for more efficient event dispatch
bufferlist: add function to add a list of buffers
clock: fix racy shutdown clock id leak
element: add support for arbitrary element class / factory details
element: link_many should activate pads if needed
gst: add math-compat.h header
datetime: add GstDateTime API
elementfactory: add utility functions to filter features by type
plugin: load the gst-python plugin loader with G_MODULE_BIND_LAZY
query: add buffering ranges API to retrieve informations about the areas of the stream currently buffered
value: add int64 range type
info: write debugging output to file if GST_DEBUG_FILE environment variable is set
pad: use more efficient g_object_notify_by_pspec() for caps notifies if compiling against new-enough GLib
pipeline: If the currently used clock gets lost update it the next time when going from PAUSED to playing
plugin: add release datetime field to GstPluginDesc and set it if GST_PACKAGE_RELEASE_DATETIME is defined
utils: speed up pad linking utility functions by not trying pads that will never work
adapter: add function to get a list of buffers; support 0-sized buffers
adapter: optimize gst_adapter_take() and gst_adapter_peek() a little
basesink: only answer the SEGMENT query in pull mode
basesrc: return values in stream time for the POSITION query
basetransform: allow the subclass to add new fields to caps when getting new caps from downstream
basetransform: avoid useless memcpy
basetransform: upstream caps-renegotiation fixes
bitreader: add inlined and unchecked versions of the most important functions
bytewriter: add inline and unchecked variants of all important functions
bytewriter: fix possible infinite loop caused by an overflow
queue: add "silent" property to suppress signal emission (for better performance)
queue: avoid unnecessary g_cond_signal() (for better performance)
queue: push newsegment event when linking in PLAYING
queue2: extend ring buffer to support RAM mode
queue2: in download mode, prevent range corruption due to race
queue2: don't send seeks beyond the end of the file upstream in pull mode (fixes apple trailers and youtube/html5 playback in webkit)
multiqueue: flush the data queue if downstream return WRONG_STATE too
gst-inspect: print GST_PARAM_MUTABLE_property flags
Bugs fixed since 0.10.30:
396774 : Make GstElementDetails extensible
482147 : [queue] Issue with current time level if source task is not started
579127 : gst-launch: disable CLOCK_LOST message handling
594504 : Need a GType of " Date AND Time AND Timezone "
600004 : underrun signal emits when i tested queue overrun test case from file /gstreamerXXXX/tests/check/element/queue.c
610366 : [gstcollectpads][doc] Add a reminder for 'data' doc
611918 : leaky queue might not push newsegment event
618919 : Registry/Plugin Loading Memory Leak
619522 : basetransform fix for upstream caps-renegotiation
621299 : make simple queues faster
621332 : BaseTransform should disable proxy alloc if downstream changes caps
622740 : GstPad: Do not call gst_pad_accept_caps() when caps change
623040 : Add release_datetime field to GstPluginDesc
623121 : [queue2] downloaded areas of the media are not exposed
623491 : make *_get_type() thread safe
623541 : [basetransform] Implement POSITION query
623622 : [basesink/basesrc] Should return values in stream time for POSITION query
623806 : [pipeline] Doesn't update the clock if the currently used one gets lost and the start time did not change
623875 : gstregistrybinary.c compatibility with glib > = 2.25.0
624203 : gstutils: Make gst_pad_proxy_getcaps() return empty caps if it's what the other side has
625239 : FTBFS: ./gstreamer-decl.txt:9461: warning: GstTagList has multiple definitions.
625295 : [info] regression: doesn't flush output stream after every debug print any longer
625368 : gstdatetime.c doesn't compile in VS 2008
625862 : [docs] unused symbol GST_CAT_LEVEL_LOG_valist breaks the build
626027 : [tag] Add GST_TAG_APPLICATION_NAME
626181 : GstElementFactory: add listing/filtering API
626651 : [tag] Photography/capture settings tags
626784 : element: link_many might assert elements are in paused or playing
627438 : gst: Add a gst_is_initialized() API
627826 : GstInt64Range type
627910 : Warnings emitted when -Wcast-qual used
627959 : [queue2] on-disk buffering failing for AVI container
628014 : Deprecate GST_FLOW_IS_FATAL/GST_FLOW_IS_SUCCESS
628174 : New gstvalue checks cause trouble in thoggen
628176 : [basetransform] Problems with buffer handling in inplace mode
628408 : Use GDateTime that has been released
629241 : Build broken with introspection using gobject-introspection from master
629410 : GstBaseTransform: position query refers to sink pad, not source pad
629494 : Latest gst-launch.c doesn't build in Visual Studio 2008
629553 : GstAdapter: timestamp not updated when empty buffer is pushed
629831 : [API] add gst_structure_take_value() and gst_structure_id_take_value()
629946 : Enumerations have incorrect names of enum values (GEnumValue.value_name)
630257 : GST_DEBUG_DUMP_DOT_DIR not working anymore
630436 : basesink: renderdelay needs to be subtracted in adjust_time()
630437 : basetransform: Make a WARNING into a DEBUG statement
630439 : clock: fix racy shutdown clock id leak
631755 : Fix build with glib 2.21.3
631853 : [queue2] deadlock when using temp-location and dispatch-properties
632236 : [gst-inspect] unhelpful uri handler output
632433 : [basesink] hangs/drops going to PLAYING following flushing step in PAUSED
632977 : [queue2] qtdemux causes soup to request seeks past the end of the range
633147 : Simple reverse negotiation pipeline is broken.
633886 : Visual Studio emits warnings about double defined _USE_MATH_DEFINES
635031 : [datetime] Fix unix epoch handling
635389 : Include information on exported packages in GIRs
635869 : GST_BOILERPLATE_FULL causes warnings in user C++ code
633176 : recent multiqueue changes break DVD playback
API additions since 0.10.30:
gst_is_initialized
gst_buffer_list_iterator_add_list
GstBin:message-forward
GST_TYPE_DATE_TIME
gst_date_time_get_day
gst_date_time_get_hour
gst_date_time_get_microsecond
gst_date_time_get_minute
gst_date_time_get_month
gst_date_time_get_second
gst_date_time_get_time_zone_offset
gst_date_time_get_type
gst_date_time_get_year
gst_date_time_new
gst_date_time_new_from_unix_epoch_local_time
gst_date_time_new_from_unix_epoch_utc
gst_date_time_new_local_time
gst_date_time_new_now_local_time
gst_date_time_new_now_utc
gst_date_time_ref
gst_date_time_unref
gst_tag_list_get_date_time
gst_tag_list_get_date_time_index
GST_TAG_GEO_LOCATION_HORIZONTAL_ERROR
GST_TAG_APPLICATION_DATA
GST_TAG_APPLICATION_NAME
GST_TAG_DATE_TIME
GST_ELEMENT_IS_SOURCE
gst_element_class_set_documentation_uri
gst_element_class_set_icon_name
gst_element_factory_get_documentation_uri
gst_element_factory_get_icon_name
gst_element_factory_list_filter
gst_element_factory_list_get_elements
gst_element_factory_list_is_type
GstElementFactoryListType
GST_ELEMENT_FACTORY_TYPE_ANY
GST_ELEMENT_FACTORY_TYPE_AUDIOVIDEO_SINKS
GST_ELEMENT_FACTORY_TYPE_AUDIO_ENCODER
GST_ELEMENT_FACTORY_TYPE_DECODABLE
GST_ELEMENT_FACTORY_TYPE_DECODER
GST_ELEMENT_FACTORY_TYPE_DEMUXER
GST_ELEMENT_FACTORY_TYPE_DEPAYLOADER
GST_ELEMENT_FACTORY_TYPE_ENCODER
GST_ELEMENT_FACTORY_TYPE_FORMATTER
GST_ELEMENT_FACTORY_TYPE_MAX_ELEMENTS
GST_ELEMENT_FACTORY_TYPE_MEDIA_AUDIO
GST_ELEMENT_FACTORY_TYPE_MEDIA_IMAGE
GST_ELEMENT_FACTORY_TYPE_MEDIA_METADATA
GST_ELEMENT_FACTORY_TYPE_MEDIA_SUBTITLE
GST_ELEMENT_FACTORY_TYPE_MEDIA_VIDEO
GST_ELEMENT_FACTORY_TYPE_MEDIA_ANY
GST_ELEMENT_FACTORY_TYPE_MUXER
GST_ELEMENT_FACTORY_TYPE_PARSER
GST_ELEMENT_FACTORY_TYPE_PAYLOADER
GST_ELEMENT_FACTORY_TYPE_SINK
GST_ELEMENT_FACTORY_TYPE_SRC
GST_ELEMENT_FACTORY_TYPE_VIDEO_ENCODER
GST_ELEMENT_FACTORY_KLASS_DECODER
GST_ELEMENT_FACTORY_KLASS_ENCODER
GST_ELEMENT_FACTORY_KLASS_SINK
GST_ELEMENT_FACTORY_KLASS_SRC
GST_ELEMENT_FACTORY_KLASS_MUXER
GST_ELEMENT_FACTORY_KLASS_DEMUXER
GST_ELEMENT_FACTORY_KLASS_PARSER
GST_ELEMENT_FACTORY_KLASS_PAYLOADER
GST_ELEMENT_FACTORY_KLASS_DEPAYLOADER
GST_ELEMENT_FACTORY_KLASS_FORMATTER
GST_ELEMENT_FACTORY_KLASS_MEDIA_VIDEO
GST_ELEMENT_FACTORY_KLASS_MEDIA_AUDIO
GST_ELEMENT_FACTORY_KLASS_MEDIA_IMAGE
GST_ELEMENT_FACTORY_KLASS_MEDIA_SUBTITLE
GST_ELEMENT_FACTORY_KLASS_MEDIA_METADATA
gst_plugin_feature_list_debug
gst_plugin_feature_rank_compare_func
gst_query_add_buffering_range
gst_query_get_n_buffering_ranges
gst_query_parse_nth_buffering_range
gst_structure_get_date_time
gst_structure_id_take_value
gst_structure_take_value
GST_TYPE_INT64_RANGE
gst_int64_range_get_type
gst_util_fraction_compare
gst_value_get_int64_range_max
gst_value_get_int64_range_min
gst_value_set_int64_range
GST_VALUE_HOLDS_DATE_TIME
gst_adapter_take_list
gst_bit_reader_skip_unchecked
gst_bit_reader_skip_to_byte_unchecked
gst_bit_reader_get_bits_uint16_unchecked
gst_bit_reader_get_bits_uint32_unchecked
gst_bit_reader_get_bits_uint64_unchecked
gst_bit_reader_get_bits_uint8_unchecked
gst_bit_reader_peek_bits_uint16_unchecked
gst_bit_reader_peek_bits_uint32_unchecked
gst_bit_reader_peek_bits_uint64_unchecked
gst_bit_reader_peek_bits_uint8_unchecked
gst_byte_writer_put_int8_unchecked
gst_byte_writer_put_int16_be_unchecked
gst_byte_writer_put_int16_le_unchecked
gst_byte_writer_put_int24_be_unchecked
gst_byte_writer_put_int32_be_unchecked
gst_byte_writer_put_int32_le_unchecked
gst_byte_writer_put_int64_be_unchecked
gst_byte_writer_put_int64_le_unchecked
gst_byte_writer_put_uint8_unchecked
gst_byte_writer_put_uint16_be_unchecked
gst_byte_writer_put_uint16_le_unchecked
gst_byte_writer_put_uint24_be_unchecked
gst_byte_writer_put_uint24_le_unchecked
gst_byte_writer_put_uint32_be_unchecked
gst_byte_writer_put_uint32_le_unchecked
gst_byte_writer_put_uint64_be_unchecked
gst_byte_writer_put_uint64_le_unchecked
gst_byte_writer_put_float32_be_unchecked
gst_byte_writer_put_float32_le_unchecked
gst_byte_writer_put_float64_be_unchecked
gst_byte_writer_put_float64_le_unchecked
gst_byte_writer_put_data_unchecked
gst_byte_writer_fill_unchecked
API deprecated since 0.10.30:
GST_FLOW_IS_FATAL
GST_FLOW_IS_SUCCESS

GStreamer Changelog

What's new in GStreamer 1.16.0

New in GStreamer 1.14.1 (May 18, 2018)

New in GStreamer 1.14.0 (Mar 20, 2018)

New in GStreamer 1.12.0 (May 4, 2017)

New in GStreamer 1.10.0 (Nov 1, 2016)

New in GStreamer 1.8.1 (Apr 20, 2016)

New in GStreamer 1.8.0 (Mar 24, 2016)

New in GStreamer 1.6.3 (Jan 20, 2016)

New in GStreamer 1.6.2 (Dec 14, 2015)

New in GStreamer 1.6.1 (Oct 31, 2015)

New in GStreamer 1.6.0 (Sep 26, 2015)

New in GStreamer 1.5.2 (Jun 26, 2015)

New in GStreamer 1.4.5 (Dec 21, 2014)

New in GStreamer 1.4.4 (Nov 10, 2014)

New in GStreamer 1.4.1 (Aug 28, 2014)

New in GStreamer 1.4 RC2 (Jul 12, 2014)

New in GStreamer 1.4 RC1 (Jun 30, 2014)

New in GStreamer 1.2.4 (Apr 22, 2014)

New in GStreamer 1.2.2 (Dec 28, 2013)

New in GStreamer 1.2.0 (Sep 25, 2013)

New in GStreamer 1.1.4 (Aug 29, 2013)

New in GStreamer 1.0.8 (Jul 12, 2013)

New in GStreamer 1.0.6 (Mar 23, 2013)

New in GStreamer 1.0.5 (Jan 8, 2013)

New in GStreamer 1.0.4 (Dec 19, 2012)

New in GStreamer 1.0.3 (Nov 21, 2012)

New in GStreamer 1.0.2 (Oct 25, 2012)

New in GStreamer 1.0.1 (Oct 8, 2012)

New in GStreamer 1.0.0 (Sep 25, 2012)

New in GStreamer 0.10.31 (Dec 2, 2010)

New in GStreamer 0.10.29 (Apr 28, 2010)