GStreamer Changelog

New in version 1.5.2

June 26th, 2015
  • 740502 : Add absolute property to GstDirectControlBinding
  • 740575 : Fixing DTS in GStreamer
  • 745366 : concat: Forward FLUSH_START / FLUSH_STOP events
  • 746949 : concat: Add active-pad property
  • 750027 : concat: Reset internal start offset to 0 after flushing seek
  • 750033 : basetransform - allow collation/separation of buffers
  • 750039 : Keeping buffers with shared memory alive
  • 750319 : memory: subclasses don't know map flags in unmap
  • 750530 : ptp: FreeBSD, DragonFly and other BSDs don't have ifreq.ifw_hwaddr
  • 750574 : netclientclock: Make the clock a wrapper clock around an internal clock
  • 750761 : inputselector: Handle different duration track selection
  • 750782 : pipeline: Add gst_pipeline_set_latency(), getter and GObject property
  • 751026 : basesink: Properly handle buffer lists for the last-sample property
  • 751047 : concat: Add adjust-base property
  • 751107 : concat: when releasing pad, send EOS appropriately.
  • 751235 : utils: get_compatible_pad does not fully respect filter caps
  • 751420 : basesink: need to deep-copy last buffer list in drain

New in version 1.4.5 (December 21st, 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 version 1.4.4 (November 10th, 2014)

  • Bugs fixed in this release:
  • 737498 : multiqueue: doesn't take GAP event into account when calculating current level
  • 737794 : multiqueue: deadlock if queue overruns with serialized events
  • 737999 : systemclock: multi-thread entry status issue
  • 738198 : multiqueue: Does not wake up not-linked streams on EOS

New in version 1.4.1 (August 28th, 2014)

  • The 1.4 release series is adding new features on top of the 1.2 series and is part of the API and ABI-stable 1.x release series of the GStreamer multimedia framework that contains new features. The 1.4.x bugfix releases only contain important bugfixes compared to 1.4.0.

New in version 1.4 RC2 (July 12th, 2014)

  • The GStreamer team is pleased to announce the second release candidate of the stable 1.4 release series. The 1.4 release series is adding new features on top of the 1.0 and 1.2 series and is part of the API and ABI-stable 1.x release series of the GStreamer multimedia framework.
  • This release candidate will hopefully shortly be followed by the stable 1.4.0 release if no bigger regressions or bigger issues are detected, and enough testing of the release candidate happened. The new API that was added during the 1.3 release series is not expected to change anymore at this point.

New in version 1.4 RC1 (June 30th, 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 version 1.2.4 (April 22nd, 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 version 1.2.2 (December 28th, 2013)

  • The 1.2 release series is adding new features on top of the 1.0 series and is part of the API and ABI-stable 1.x release series of the GStreamer multimedia framework that contains new features.

New in version 1.2.0 (September 25th, 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 version 1.1.4 (August 29th, 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 version 1.0.8 (July 12th, 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 version 1.0.6 (March 23rd, 2013)

  • build fixes for out-of-tree autogen.sh and automake 1.13, and recent kernel/video4linux
  • gobject-introspection fixes for bindings
  • videoscale: Correct DAR and border calculations
  • playbin: make sure converters are always plugged when needed, fixes not-negotiated errors with some sinks
  • playbin: fix subtitleoverlay caps handling to avoid not-negotiated errors when subtitle plugins are missing
  • adder: make "caps" property work properly
  • alsasink: don't use 100% CPU in some cases
  • reliability fixes for flushing seeks and shutdowns in queue and bastransform
  • appsrc: fix locking order
  • audiovisualisation fixes
  • deinterlace: fix infinite loop on EOS with non-default methods or fields
  • avidemux push mode fixes, make dv-in-avidemux work
  • level: send a final level message on EOS
  • osxvideosink fixes
  • ximagesrc: Set the pixel aspect ratio correctly in the output caps
  • v4l2: don't check stride for encoded formats
  • leak fixes in GstBin, pango, auparse, gdppay
  • qtdemux: skip disabled tracks and ignore chapter subtitle tracks
  • RTSP and RTP fixes
  • Opus audio decoder, encoder and RTP payloader fixes
  • codecparser fixes for H.264, MPEG-2 and VC-1 parsing
  • opensles, eglglessink, and decklink plugins ported to 1.0
  • libav: fix H.264 decoding errors in some files and update to 0.8.5 release
  • miscellaneous bug fixes

New in version 1.0.5 (January 8th, 2013)

  • rtspsrc: fix regression that make rtspsrc hang when stopping
  • alsasrc: don't output buffers without timestamps or with bogus timestamps
  • add GST_CLOCK_TYPE_OTHER clock type
  • discoverer, decodebin: fix state change re-sync race that might lead to deadlocks
  • add GST_BIN_FLAG_NO_RESYNC flag that disables a resync when an element is added, removed or linked in the bin; this is interesting for complex bins that dynamically add elements to themselves and want to manage the state of those elements without interference from state resync threads (which may cause deadlocks)
  • video: fix crashes with and frame sizes of A420 video format
  • fix VP6-alpha video playback
  • jpegenc: pass flow returns upstream
  • audio/video parsers: fix negotiation with encoders in some transcoding cases
  • qtdemux: fix pixel-aspect-ratio of some files with ProRes video
  • cairo: port cairooverlay to 1.0
  • psdemux: tentative port to 1.0; take into account both DTS and PTS
  • shm: Actually get the permissions on get_property
  • waylandsink: do not default to fullscreen mode
  • build: fix build with new automake 1.13
  • build: pass CC, LD, AS, AR and NM to the libav configure if set
  • miscellaneous bug fixes

New in version 1.0.4 (December 19th, 2012)

  • basesrc: fix potential leaks when re-activating in a different mode
  • bindings: make all pad probe types work with bindings
  • bindings: fix gst_event_parse_stream_start() annotation, fixing crash
  • bindings: fix annotation for gst_app_src_push_buffer(), fixing crash
  • bindings: add several missing annotations for GstRtspMessage API
  • documentation improvements
  • playbin: fix occasional not-negotiated errors when switching visualisations
  • ssaparse: ignore invalid UTF-8 in SSA/ASS subtitles init sections in matroska files
  • streamsynchronizer: better timestamp and gap handling at EOS, fixing potential OOM in baseaudiosink
  • deinterleave: properly set srcpad channel position
  • osxvideosink: Fix resizing the Cocoa window on receiving new caps
  • rtspsrc fixes
  • shout2send: also accept audio/webm in addition to video/webm
  • videobox: fix border filling for planar YUV formats
  • webmmux: fix linking to shout2send
  • v4l2: fix build on FreeBSD
  • siddec: initialize debug category
  • mpeg4videoparse: also parse divx 4/5
  • mpeg4videoparse: export number of sprite warping points in caps (decoders might have no or only limited GMC support)
  • mpegtsmux: propagate flow returns upstream; don't crash when reused
  • rtmpsrc: disable seeking if the configured url specifies live=true
  • build fixes for OS/X (shm) and Windows (d3dvideosink)
  • libav G.726 decoder fixes
  • miscellaneous bug fixes
  • some memory leak fixes

New in version 1.0.3 (November 21st, 2012)

  • bufferpool: fix deadlock
  • baseparse: forward stream-start event in push mode, fixing issues with streamsynchronizer
  • basesink: reset START_TIME when needed, fixing position reporting after seeking beyond end
  • typefind: detect isml ftyp as iso-fragmented video/quicktime
  • typefinding improvements fixing playback of some wavpack files
  • textoverlay rendering fixes
  • gobject-introspection annotation fixes
  • API: gst_video_decoder_get_qos_proportion()
  • rtspsrc: numerous improvements
  • build fix for gst-plugins-base installed in non-default prefix
  • multifilesink: post messages in max-size mode as well
  • vp8dec: improve robustness on decoding errors, e.g. for videocalling
  • cdio: try to handle CD-TEXT in non-UTF8 encodings
  • xingmux, siddec, dvdlpcmdec and dvdsubdec fixes
  • mpegtsmux: fix DTS/PTS confusion
  • tsdemux, tsparse: seeking fixes
  • tsdemux, tsparse: fix timestamping with push-based input
  • h264parse: fix PPS insertion
  • mpg123audiodec: fix leaks from not chaining up in the finalize function
  • avcodecmap: Y41B is YUV411P, not YUV410P
  • numerous bug fixes
  • some memory leak fixes

New in version 1.0.2 (October 25th, 2012)

  • Changes since 1.0.0:
  • Capsfilter prefers filter caps over passthrough now
  • Application Development Manual, Plugin Writer's Guide and other documentation updated and extended for 1.0
  • Bug fixes
  • Bugs fixed since 1.0.0:
  • 680862 : identity with single-segment=true gives buffer.pts of CLOCK_TIME_NONE
  • 684538 : baseparse: no timestamps after seeking in mp3 or aac
  • 684755 : typo - whithin > within
  • 684765 : Plugins without a klass in the metadata crashes autoaudiosink
  • 684809 : proxypad don't hold a ref to their internal pad while streaming through it
  • 684970 : Don't register printf extension for %p when glib is not using system printf
  • 684981 : Pipeline hangs on PREROLLING negotiating caps
  • 685072 : memory: map(READ)/unmap clears the READONLY status
  • API additions since 1.0.0:
  • gst_base_transform_set_prefer_passthrough()

New in version 1.0.1 (October 8th, 2012)

  • This is a new bug-fix release for the new API and ABI-stable 1.x series of the GStreamer multimedia framework.

New in version 1.0.0 (September 25th, 2012)

  • more flexible memory handling
  • extensible and negotiable metadata for buffers
  • caps negotiation and renegotiation mechanisms, decoupled from buffer allocation
  • improved caps renegotiation
  • automatic re-sending of state for dynamic pipelines
  • reworked and more fine-grained pad probing
  • simpler and more descriptive audio and video caps
  • more efficient allocation of buffers, events and other mini objects
  • improved timestamp handling
  • support for gobject-inspection-based language bindings
  • countless other improvements

New in version 0.10.31 (December 2nd, 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

New in version 0.10.29 (April 28th, 2010)

  • improve plugin loading robustness: do not ever unload a plugin after calling into it: should fix mystery crashers during registry loading when a plugin init function returns FALSE (e.g. when some supporting library fails to initialise or a wrapper plugin found no features to wrap and wrongly returned FALSE)
  • configurable memory alignment for GstBuffers
  • add QoS message to inform apps of lost data, dropped frames etc.
  • basesink, basetransform: add support for new QoS message
  • basetransform: accept non-fixed caps suggestions
  • basesrc: fix gst_base_src_new_seamless_segment()
  • GstController fixes and optimisations
  • set thread name for pad tasks on Linux
  • pipeline, bin: fix refcount issue when removing elements during a state change
  • queue2: implement seeking in download mode
  • queue2: implement flushing in download buffering
  • queue2: improve buffer level measurement in download mode
  • fdsrc: allow specifying the size in bytes on the uri
  • build fixes: better checks for uint128_t, inline assembly on OSX, compilation if HAVE_REGISTER_PRINTF_SPECIFIER is undefined, gobject-introspection
  • two symbols were removed that had been exported but never been used or been declared in any header file: gst_element_default_error and gst_element_request_compatible_pad