-This is GStreamer Good Plug-ins 0.10.6, "Wobble Board"
-
-Changes since 0.10.5:
-
- * Much improved RTSP/RTP and V4l2 support
- * New plugins - audiopanorama, audioinvert, audiodynamic, audioamplify
- * OSX and Windows video/audio support elements moved from Bad Plugins
- * qtdemux, videocrop and wavpack elements moved from Bad Plugins
- * Fixes in avi and matroska muxing
- * Fixes in wavparse, sunaudio, AVI demuxing, ID3 tag handling
- * gamma element ported to 0.10
- * Parallel installability with 0.8.x series
- * Threadsafe design and API
-
-Bugs fixed since 0.10.5:
-
- * 392189 : [esdsink] pipeline hung in state change from PAUSED to PL...
- * 407590 : crash using gconfaudiosink
- * 439255 : [rtspsrc] crash on unsupported transport
- * 441118 : Incorrect caps in G.711 RTP depayloaders
- * 323471 : [PLUGIN-MOVE] osxaudio
- * 407780 : avimux does not handle video/x-h264
- * 316203 : MinGW: udp and rtsp plugin should be disabled on windows
- * 340362 : [PATCH] new plugin - hardlimiter
- * 342463 : [ximagesrc] performance improvement
- * 347806 : [v4l2src] try_capture(): Error getting buffer information...
- * 350296 : [udpsrc] add property to remove extra headers from buffers
- * 354007 : [PLUGIN-MOVE] videocrop should be moved to -good
- * 356692 : wavparse drops final sample in most files
- * 358040 : Fix speex rtp (de)payloader
- * 362566 : [dv1394src] Random segfault and kernel oopses
- * 374489 : rtspdec never sends out RTCP Receiver reports
- * 377306 : [sunaudiomixer] mixer track labels not localized
- * 379298 : [avimux] allow multiple audio streams
- * 392620 : gnome-sound-record can not be started on sunray client on...
- * 393427 : error/warnings when compiling with mingw
- * 395688 : playbin is unable to play rtsp stream for 3gp from Darwin...
- * 396057 : [audiofx] New audioinvert element
- * 397162 : [audiofx] New element audioamplify
- * 397946 : [audiopanorama] another trivial docs fix
- * 398299 : [smpte] crashes if fed empty buffers from fakesrc
- * 398325 : [smpte] Segfaults with big width/height
- * 399338 : Hang in PREROLLING when trying to play a specially crafte...
- * 399825 : Tags don't work properly for shout2send plugin
- * 406042 : [v4l2src] fails with radio chips
- * 407369 : [v4l2src] Wrong way of EIO error handling
- * 407793 : memory leaks of sdpmessage.c
- * 407797 : bug of rtsp_url_parse() in gst/rtsp/rtspurl.c
- * 408544 : totem crashed with SIGSEGV to rtsp_ext_wms_configure_stre...
- * 410997 : Fails to build with -z defs
- * 412597 : Simplify GstSunAudioSrc
- * 412608 : [avidemux] flow return aggregation can ignore errors
- * 414168 : Mixes spaces and tabs in Makefile.am defs
- * 416445 : [avidemux] seeking regressions
- * 416727 : [v4l2src] typo in printf format string
- * 416728 : [v4l2src] typo in translated strings
- * 417729 : [autoaudiosink] plugs alsaspdifsink, breaking playbin
- * 419338 : [wavparse] critical warning from plugin viewer
- * 420208 : Probably typo
- * 426044 : [flacenc] broken files when used with oggmux
- * 427990 : [rtph264depay] sprop-parameter-set erros
- * 428611 : [ximagesrc] segfaults when used from a non-X terminal
- * 428901 : [icydemux] broken tags for non-UTF-8/ISO-8859-1-using ra...
- * 429319 : [alphacolor] distorts png images without alpha channel
- * 429666 : [goom] totem skips mp3 (while Rhythmbox doesn't)
- * 430228 : [sunaudio] copyright bug
- * 430632 : memory problem
- * 430804 : [PATCH] navseek only seeks (sort-of) partially
- * 431282 : broken RTP depayloaders
- * 433119 : wavparse causes skipping for MPEG-encoded RIFF Wav files ...
- * 433135 : [wavparse] regression in CVS with mp3-in-wav
- * 433530 : udpsrc read of size 0
- * 434824 : rtp mp4 payloader doesn't handle newsegments and flush ev...
- * 436910 : [videomixer] wrong strides with odd input width
- * 437499 : [wavparse] can't handle WAV file with 'LIST' header chunk
- * 437670 : Improvements for rtsptransport.[ch]
- * 437692 : Some more fixes for the RTSP support
- * 438926 : invalid comparison of pointer with string literal
- * 438940 : [rtph263ppay] Support for Segment Fragmentation based on ...
- * 440127 : Autoaudiosink does extra switching between null and ready...
- * 440203 : Support multiple RTSP headers of the same type
- * 440928 : Improved RTSP version support, and added Date header
- * 441408 : speexdec plays incorrectly files made with nframes=4
- * 442535 : [wavenc] Doesn't handle width!=depth files with audio/x-r...
- * 442677 : WideBand AMR payloaders
- * 443081 : [wavparse] fails to post error on FLOW_NOT_LINKED
- * 445905 : ximagesrc disregards display_name property
- * 446981 : error during the compilation of rtspconnection.c
- * 447210 : wrong length calculation for codec_data
- * 447458 : [qtdemux] export AMRSpecificBox as codec_data
- * 385887 : [flac] make work with libflac-1.1.3
- * 404646 : [audiofx] Compressor/Expander element
- * 424527 : [auparse] don't convert non-native endianness floats in t...
- * 387121 : [matroskamux] Can't mux raw audio
- * 392855 : [matroska] plugin must link against zlib (error with MinGW)
- * 394851 : [audiopanorama] Some trivial docs fixes
- * 394859 : [audiopanorama] New simple method for adjusting the panorama
- * 398086 : [smpte] crashes if input dimensions differ
- * 406018 : 64bit uncleanness in gstavimux.c
- * 407057 : [wavparse] leaks contents of every buffer pushed when in ...
- * 407349 : [id3demux] wrongly interprets TDAT as year
- * 350278 : [rtpmp2tdepay] Add support for proprietary headers
- * 380895 : A couple of corrections and improvements for the RTSP sup...
- * 380944 : [dvdec] Doesn't set pixel-aspect-ratio
- * 394977 : multipartmux not honoring flow return
- * 403956 : Add float32 support for " level " element
- * 405213 : mp3 plays too fast in totem or rhythmbox
- * 407006 : [goom] odd adapter behaviour
- * 412704 : [PATCH] gamma filter ported to 0.10
- * 414887 : [gconf] gconfaudiosink doesn't set GST_ELEMENT_IS_SINK un...
- * 415446 : [avidemux] fails parsing mjpeg file from digital camera
- * 417792 : rtp depayloader for AAC
- * 423304 : file descriptor closed in udpsrc and dynudpsink
- * 423782 : Code for H264 payloader
- * 428182 : Current CVS generates compiler warnings
- * 429329 : [videobox] add support for AYUV input
- * 442874 : Multipartmux assumes caps == mime
-
-Changes since 0.10.4:
-
- * Parallel installability with 0.8.x series
- * Threadsafe design and API
- * RTP/RTSP improvements
- * Fixes in OSS support
- * Addition of the audiopanorama element
- * Improvements in AVI playback
- * Annodex playback fixes
- * Support FLAC in OGG and Matroska
- * Fixes in the Speex decoder
- * V4L2 source moved from Bad Plugins
- * SMPTE element ported to 0.10
- * GStreamer Data Protocol (GDP) Payloader and Depayloader elements added
- * Many other bug-fixes
-
-Bugs fixed since 0.10.4:
-
- * 336465 : [patch] Streaming support for avidemuxer
- * 349207 : [PLUGIN-ADD] audiopanorama
- * 341278 : [autoaudiosink] should fallback to fakesink
- * 342950 : Implement device profiles in autoaudiosink/autoaudiosrc
- * 348233 : dv1394src crashes with libavc1394 0.5.3 - cause known
- * 349015 : [sunaudio] open source with O_NONBLOCK
- * 349894 : RTSP Multicast
- * 351347 : --disable-schemas-install now works too well
- * 351794 : [id3demux] try harder to extract wrongly marked strings
- * 352577 : [avidemux] regression in CVS with Elephant's Dream
- * 355210 : Sample pipeline from the documentation doesn't work properly
- * 356142 : GST Sun Audio Mixer doesn't set only Output Track as Mast...
- * 356147 : [avimux] duration in header not correct for big avi ( > 2 ...
- * 357592 : Avoid compiler warnings with uClibc and -Werror
- * 361637 : h263 variant missing from RTP (de)payloaders' caps
- * 361639 : MPA payloader's payload number is incorrect
- * 362603 : Fixes compiling with forte: warning clean up (part 4)
- * 362673 : Playback with 4Front OSS driver not working due to blocki...
- * 366492 : add windows vs8 project files
- * 369621 : [avidemux] Out-of-sync playback with VBR MP3 audio
- * 372021 : flxdec has wrong classification
- * 374213 : Seeking with LADSPA plug-ins fails
- * 374479 : [PATCH] videomixer memleak fix and enhancement
- * 374737 : [matroskademux] doesn't recognise opaque " A_AAC " codec ID
- * 376594 : id3demux crashes when reading compressed ID3 frames
- * 379433 : [PATCH] avidemux audio pad reports wrong position upon query
- * 379792 : Remove memcpy in multipartmux and fix RFC compliance
- * 379918 : Doesn't compile with newer libcaca versions (0.99.beta4+)
- * 380199 : [matroskademux] Wrong framerate conversion
- * 380825 : make avimux accept video/mpeg in versions 1, 2 and 4
- * 381857 : [id3v2mux] crashes trying to write empty frames
- * 382179 : Videomixer shouldn't reset position to 0 when the caps ar...
- * 382277 : multipartmux modifies buffer timestamp
- * 382982 : [apedemux] Fails to read track gain or other doubles
- * 383001 : [PATCH] if using multicast in udpsrc, bind to the multica...
- * 383043 : Videomixer can crash when adding/removing pads while pla...
- * 383596 : mixer doesnot work if set AUDIODEV on solaris
- * 384587 : libcaca check breaks cross-compile
- * 385031 : [id3demux] autoplug loop if first buffer has nonzero offs...
- * 385623 : [quarktv] crash when plugged dynamically into a pipeline
- * 343348 : [matroska] add support for vobsub subtitles
- * 345449 : [icydemux] Internet radio hangs when connecting to radio....
- * 348762 : [ID3v2] comment frame extraction/writing doesn't retain d...
- * 351116 : 'make check' tries to run annodex unit tests even if anno...
- * 352110 : [flacdec] no support for ogg flac
- * 366155 : [matroskademux] Several problems in encoding handling code
- * 374658 : [matroskamux] add tag writing support and implement relea...
- * 378184 : g-s-p crash due to incorrect free from do_toggle_element
- * 360673 : [PATCH] Stuttering with SunAudio Sink
- * 349068 : multipart demuxer improvements
- * 350006 : [LADSPA] after changing state from PLAYING to READY/NULL,...
- * 350340 : CMML test cases and small fixes
- * 350785 : [ossmixer] provides no way to set mixer device
- * 353908 : Error seeking at the beginning of a CMML file
- * 356596 : [wavparse] Does not support GSM encoded audio
- * 358156 : in udpsrc.c gst_udpsrc_create function read command could...
- * 361252 : Memory leak in udpsrc
- * 364958 : [avidemux] broken timestamping in streaming mode
- * 367221 : [PATCH] videomixer does not mix for some framerate combin...
- * 368162 : iLBc rtp payloaders and depayloaders not compiled
- * 377792 : rtspsrc tries to stream application/x-wms-rtx
- * 383323 : smpte doesn't handle sink1 and sink2 correctly
- * 383726 : [audiopanorama] only transforms half of the samples in fl...
- * 349901 : [LADSPA] gst_element_class_add_pad_template: assertion `g...
- * 375476 : v4l2src cannot close /dev/video0
-
-Changes since 0.10.3:
-
- * added apev2mux element
- * use libiec61883 for Firewire
-
-Bugs fixed since 0.10.3:
-
- * 345930 : [id3demux] segfaults with file containing only ID3v1 tag
- * 347529 : wavparse error
- * 321191 : rtpamrdec isn't a subclass of GstBaseRtpDepayload
- * 340027 : [patch] wavparse fails for several files
- * 318563 : offer support for new raw1394_iso api in dv1394src
- * 330623 : [avidemux] only the beginning of big avi files gets played
- * 334375 : [id3demux] [id3v2mux] ID3 tag rewriting is lossy
- * 337076 : Problem with broken matroska files containing non-UTF8 su...
- * 339704 : [id3demux] read images from ID3 tags
- * 340282 : Goom visualization is unusable at 'Normal' size and higher
- * 340623 : [matroskademux] small memory leak
- * 340699 : [flacdec] should not send EOS when doing segment seeking
- * 340859 : [avimux] produces index with all frames marked as keyframes
- * 340946 : raw1394 plugin uses deprecated functions
- * 340979 : [id3demux] mp3 id3v2 TCON tag possible bug?
- * 341489 : gst-plugins-good wavparse Cygwin fix
- * 341774 : Fails to read tags in file
- * 341818 : [matroskademux] poor concurrent performance
- * 342029 : [id3demux] overflow of titlenumbers
- * 342097 : [jpegdec] crash with attached JPEG file
- * 342448 : [matroska] support for muxing/demuxing Theora video
- * 342526 : [avimux] dml index support, codec_data support, cleanups
- * 342592 : dvdemux doesn't post segment-done right
- * 342734 : [matroskamux] might block on state-change
- * 343051 : [autoaudiosink] doesn't try esdsink
- * 343055 : README mentions 0.9.6
- * 343117 : jpegdec, mjpeg avi's and flush seeks
- * 343122 : [taglib] new apev2mux element
- * 343123 : [apedemux] add support for GST_TAG_LOCATION
- * 343127 : [apedemux] extract track count, clean up parsing
- * 343602 : configure --disable-external fails
- * 343603 : need to add -lm to build tests/examples/level
- * 343678 : configure.ac incorrectly uses $(SED) rather than $SED
- * 343837 : [wavparse] can't handle WAV file with 'bext' header chunk
- * 344100 : --disable-schemas-install not honored
- * 344101 : SunAudio mixer fixes
- * 344120 : dv1394src should now require libraw1394 > = 1.1.0
- * 344605 : [id3demux] set picture type on image buffers
- * 345232 : [wavparse] reads beyond end-of-file (in pull mode)
- * 345288 : [udp] make work on Windows
- * 345713 : ximagesrc uses XFixesCursorImage incorrectly and will seg...
- * 346066 : [sunaudiosink] don't override user setting and switch on ...
- * 346259 : [sunaudio] move monitor to input tab in mixer
- * 347234 : streaming UDP (MPEGTS) shows only one frame of video
- * 347258 : [wavparse] internal stream error reading gnome-game gnibb...
- * 347898 : [id3v2mux] write GST_TAG_ENCODER and GST_TAG_ENCODER_VERSION
- * 347972 : [cdiocddasrc] core dumps if device is not found
- * 348644 : [id3demux] Gets the wrong part of binary blob for ID3 v2....
- * 348752 : [udpsrc] add property to set buffer size for udp socket
- * 348913 : [id3v2mux] tagging utf-8 text may be converted to iso-8869-1
- * 349155 : [smokeenc] does not set caps on it's buffers
- * 349189 : LADSPA gstsignalprocessor.c: line 408: assertion failed: ...
- * 349907 : multiudpsink messes up multicast addresses
- * 350433 : [rtph263pdepay] h.263plus depayloader does not work
- * 340492 : [flacdec] support push-based operation (and thus flac-ove...
- * 345679 : fix to avoid goom core dumping
- * 317470 : [GstCheck] gst_check_teardown_element asserts wrong refcount
- * 340980 : [pixbuf,wavparse] fix build with gcc 2.95
- * 343661 : Jpeg image crashes gstreamer
- * 344923 : New SunAudio source plugin, and mixer now supports stereo...
- * 345301 : [PATCH] gst-plugins-good rtsp for Windows
- * 346921 : gstmultiudpsink multicast support is broken
- * 344136 : More accurate list of plugins which will/will not be buil...
-
-Changes since 0.10.2:
-
- * Annodex/CMML support
- * RTSP and RTP enhancements
- * HAL configured audio device support
- * FLAC, Matroska, AVI, WAV, ID3, APE, DV and JPEG plugin improvements
- * Recognise SSA/ASS and USF subtitles in Matroska files
- * Fixes for ESD and SunAudio output plugins
- * More uniform plugin descriptions
- * IceCast metadata reading plugin added
- * New plugins ported from 0.8: OSX audio, AVI muxer, X-Windows input,
- WAV encoder, Gdk-Pixbuf image decoder, Smoke decoder,
- Video colour balance
- * Lots of bug fixes
-
-Bugs fixed since 0.10.2:
-
- * 335067 : RTSP src not working with WMServer servers
- * 333657 : Replacing icy demuxing in gnomevfssrc
- * 329106 : HAL sound device wrapper plugins
- * 337749 : totem (gstreamer) crashes when playing an avi file
- * 330885 : avidemux does not handle eos at end of seek-region
- * 337364 : faulty GObject macros
- * 337625 : [patch] Streaming support for wavparse
- * 150363 : [pngdec] doesn't handle grayscale or paletted
- * 154744 : Time slider does not work with avi videos from Cannon SD100
- * 161712 : [auparse] .au files don't play in playbin
- * 313266 : [wavparse] will not play DTS stream in malformed WAV
- * 319183 : rtspsrc filter sometimes uses an odd port for rtp
- * 319986 : annodex decoding and encoding support
- * 323721 : [id3demux] read in replaygain information from RVA2 frame...
- * 323880 : " Seek in ready " for dvdemux
- * 325191 : problem with auparse or mulawdec, choppy esd playback
- * 326160 : videobalance not ported to new GstVideoFilter
- * 327658 : " Seek in ready " support for wavparse plugin
- * 328327 : gst-plugins-good fail to compile with gcc 4.1
- * 329107 : Profile support for gconfaudiosink
- * 330239 : Crash playing any song from a particular album over rhyth...
- * 330678 : Unable to play .fli files
- * 331253 : Critical warnings when using cddacdiosrc
- * 331368 : Gstreamer doesn't recognise tags
- * 331385 : [alpha] state change function returns a constant
- * 331672 : Another file that gstreamer can't read the tags on
- * 331905 : [jpeg] smokedec not ported
- * 331917 : [pngdec] does not support files with png streams
- * 332031 : [PATCH] avimux ported to 0.10
- * 332547 : [wavparse] does not support multichannel wavs
- * 333070 : [id3demux] reads unicode tags incorrectly where .8 did it...
- * 333302 : [apedemux] some WavPack files with APE tags fail to play ...
- * 333392 : [sunaudio] unused variables break CVS build with -Werror
- * 333512 : [PATCH] Fix gst_pad_new_from_template (gst_static_pad_tem...
- * 333624 : invalid get_times implementation in gstdynudpsink
- * 333784 : [patch] unref the result of gst_pad_get_parent
- * 334083 : [jpegdec] wrong durations set on buffers after seeking in...
- * 334522 : avi of mpeg4 video and adpcm audio from digital camera re...
- * 334732 : [id3demux] mp3 fails to play because typefinding thinks i...
- * 334995 : [goom] zoom filter leaked
- * 335231 : [wavparse] incorrect way to calculate seek position with ...
- * 335755 : rhythmbox import crasher - png?
- * 335858 : Video playback out of sync
- * 335958 : [speexenc] doesn't work
- * 336110 : move taglib-based ID3 muxer to -good
- * 336602 : plugins need better/univied descriptions
- * 336756 : move ximagesrc to gst-plugins-good
- * 336889 : [avidemux] index creation might fail with some non-indexe...
- * 336904 : Problem playing some AVI file when splitting large chunks...
- * 337033 : [speex] can't seek in speex-encoded audio
- * 337183 : [matroska] " caps not real subset " when playing audio files
- * 337294 : AVI files downloaded from vidoe.google.com won't play
- * 337421 : [sunaudiosink] some fixes
- * 337609 : [flacdec] can't play .flac files where header says total_...
- * 338290 : [flacdec] segment seek not supported
- * 338713 : [id3demux] TCO genre tags (id3v2.2) don't get read by gst...
- * 338715 : [shout2send] fix crash on error and tags received before ...
- * 338716 : [wavenc] " not negotiated " error with CVS core
- * 338810 : [matroskademux] blocks on segmenting seek (and other seek...
- * 339446 : [matroska] can't play file if details come before type in...
- * 339451 : [matroska] enhancement for VfW compatibility cases
- * 339678 : [matroskamux] wrong timestamps of B-frames
- * 340346 : [matroskamux] blocks upon muxing video and vorbis-audio
- * 319884 : rtpamrdec discards non-transmitted frames
-
-Changes since 0.10.1:
- * New libcdio based CDDA reading element
- * APE tag reader ported
- * ID3 tag reading fixes
- * Sun Audio Sink fixes
- * GOOM and gconf element fixes
- * lots of bug and leak fixes
-
-Bugs fixed since 0.10.1:
- * 328336 : silence warings which make dvdec / dvdemux unusable
- * 315557 : Internal event problem with MP3s from vgmix.com
- * 323327 : [cdio] port cddasrc to 0.10
- * 325148 : Bugs in G711 RTP packetization logic
- * 325649 : apetag plugin needs porting to 0.10
- * 326446 : check that all elements in -good pass queries they can't ...
- * 326602 : id3demux is not compiling without ZLIB
- * 326683 : build problem caused by AS_LIBTOOL_TAGS([CXX])
- * 326736 : gconf(audio|video)sink response to key changes
- * 326864 : [wavparse] time to bytes format conversion broken
- * 327009 : [esdsink] won't compile with includes in non-standard prefix
- * 327765 : [sunaudio] fixes for mixer and stuttering mp3 playback
- * 327825 : [matroskamux] Matroska muxer deadlock
- * 327871 : [videobox] crash when cropping
- * 328241 : id3demux emits NULL date for year tags
- * 328264 : Fix build with gcc 2.95
- * 328531 : [matroskamux] doesn't send newsegment event, critical war...
- * 329181 : totem crash when using goom effect
- * 329810 : Fails to read ID3 tag
- * 330005 : Please use the autodetect sinks by default
- * 317658 : [cdio] support for cd-text and cd-g
-
-Changes since 0.10.0:
-
- * new id3 demuxer (replaces the mad one in gst-plugins-ugly)
- * memleak fixes in avidemux, wavparse, level, smoke
- * ports of multipart,
- * fixes in flacdec, flxdec, rtp
- * documentation updates on videomixer
- * added new sunaudiosink, gconfaudiosrc and gconfvideosrc elements
-
-Bugs fixed since 0.10.0:
-
- * 321269 : add sunaudio to 0.9
- * 322769 : The ID3 tag of this file is a segfaulter
- * 323021 : sockfd property to udpsrc/dynudpsink elements
- * 322975 : erroneous audio specs in flac plugin
- * 323226 : block/crash on id3 v2 tags when using big blocksize
- * 323717 : < netinet/in.h > inclusion necessary on some systems
- * 323718 : [oss] does not build on OpenBSD 3.8 because of hardcoded ...
- * 323896 : pngdec/videomixer negotation problem in 0.10
- * 324011 : Invalid payload type definition for some rtp payloaders
- * 324012 : Invalid caps on rtpspeexpay element
- * 325504 : [flacdec] gst_flac_dec_convert_src [mis]uses g_assert
- * 325974 : [gst0.10] doesn't correctly gets the tags on a mp3
- * 326612 : Serious memory leak in level plugin
- * 326618 : memleak fix in smokeenc
-
-API added since 0.10.0:
-
- * device-name property on ossmixer subclasses
- * GstUDPSrc::sockfd property
-
-Changes since 0.9.6:
-
- * Parallel installability with 0.8.x series
- * Threadsafe design and API
- * effectv elements ported
- * videoflip updated
- * multipart ported
- * dv seeking fixed
- * rtp elements renamed
-
-Bugs fixed since 0.9.6:
-
- * 322377 : udpsrc leaks sockets
- * 322643 : Incorrect matroska frame default duration
- * 322645 : Matroska muxer: wrong pixel aspect ratio
- * 322667 : [jpegenc] leaks input buffer
- * 322794 : udp plugin linked against gstnet instead of gstnetbuffer
-
-Changes since 0.9.5:
-
- * added speex RTP payloader/depayloader
- * ported cutter
- * fractional framerates
- * more video filters now use BaseTransform
-
-Bugs fixed since 0.9.5:
-
- * 319184 : rtspsrc: invalid read in sdp_message_parse_buffer ()
- * 321001 : [matroskademux] should seek to nearest preceding index en...
- * 321430 : goom fails to register on amd64
-
-Changes since 0.9.4:
-
- * matroskamux fixes
- * wavenc fixes
- * cairotextoverlay ported
-
-Bugs fixed since 0.9.4:
-
- * 315194 : Licence information inconsistency of gst-plugins-good/gst...
- * 319731 : [matroska] SimpleBlock support for muxer and demuxer
- * 320308 : [matroska] set timestamps for buffers with ebml elements
- * 320920 : [osssink] tries to reuse a bad file descriptor
- * 321136 : [matroska-mux] avoid reading from unref'ed buffer
-
-Changes since 0.9.3:
-
- * DV/Firewire fixes
- * speexenc, cairotimeoverlay, matroska, pngdec, flxdec, videomixer,
- alphacolor ported
-
-Bugs fixed since 0.9.3:
-
- * 316204 : MinGW compilation: smtpe plugin has undefined symbols fro...
- * 316205 : Debug category for wavenc is not defined
- * 318847 : Matroska muxer port to 0.9
-
-Changes since 0.9.1:
-
- * Parallel installability with 0.8.x series
- * Threadsafe design and API
-
-Bugs fixed since 0.9.1:
-
- * 316202 : MinGW compilation: undefined autoconf macro GST_DOC
- * 317338 : [osssink] can't handle mono
+
+
+GSTREAMER 1.16 RELEASE NOTES
+
+
+GStreamer 1.16 has not been released yet. It is scheduled for release in
+March 2019.
+
+1.15.x is the unstable development version that is being developed in
+the git master branch and which will eventually result in 1.16.
+
+1.16 will be backwards-compatible to the stable 1.14, 1.12, 1.10, 1.8,
+1.6, 1.4, 1.2 and 1.0 release series.
+
+See https://gstreamer.freedesktop.org/releases/1.16/ for the latest
+version of this document.
+
+_Last updated: Wednesday 27 January 2019, 00:30 UTC (log)_
+
+
+Introduction
+
+The GStreamer team is proud to announce a new major feature release in
+the stable 1.x API series of your favourite cross-platform multimedia
+framework!
+
+As always, this release is again packed with many new features, bug
+fixes and other improvements.
+
+
+Highlights
+
+- GStreamer WebRTC stack gained support for data channels for
+ peer-to-peer communication based on SCTP, BUNDLE support, as well as
+ support for multiple TURN servers.
+
+- AV1 video codec support for Matroska and QuickTime/MP4 containers
+ and more configuration options and supported input formats for the
+ AOMedia AV1 encoder
+
+- Support for Closed Captions and other Ancillary Data in video
+
+- Support for planar (non-interleaved) raw audio
+
+- GstVideoAggregator, compositor and OpenGL mixer elements are now in
+ -base
+
+- New alternate fields interlace mode where each buffer carries a
+ single field
+
+- WebM and Matroska ContentEncryption support in the Matroska demuxer
+
+- new WebKit WPE-based web browser source element
+
+- Video4Linux: HEVC encoding and decoding, JPEG encoding, and improved
+ dmabuf import/export
+
+- Hardware-accelerated Nvidia video decoder gained support for VP8/VP9
+ decoding, whilst the encoder gained support for H.265/HEVC encoding.
+
+- Many improvements to the Intel Media SDK based hardware-accelerated
+ video decoder and encoder plugin (msdk): dmabuf import/export for
+ zero-copy integration with other components; VP9 decoding; 10-bit
+ HEVC encoding; video post-processing (vpp) support including
+ deinterlacing; and the video decoder now handles dynamic resolution
+ changes.
+
+- The ASS/SSA subtitle overlay renderer can now handle multiple
+ subtitles that overlap in time and will show them on screen
+ simultaneously
+
+- The Meson build is now feature-complete (*) and it is now the
+ recommended build system on all platforms. The Autotools build is
+ scheduled to be removed in the next cycle.
+
+- The GStreamer Rust bindings and Rust plugins module are now
+ officially part of upstream GStreamer.
+
+- Many performance improvements
+
+
+Major new features and changes
+
+Noteworthy new API
+
+- GstAggregator has a new "min-upstream-latency" property that forces
+ a minimum aggregate latency for the input branches of an aggregator.
+ This is useful for dynamic pipelines where branches with a higher
+ latency might be added later after the pipeline is already up and
+ running and where a change in the latency would be disruptive. This
+ only applies to the case where at least one of the input branches is
+ live though, it won’t force the aggregator into live mode in the
+ absence of any live inputs.
+
+- GstBaseSink gained a "processing-deadline" property and
+ setter/getter API to configure a processing deadline for live
+ pipelines. The processing deadline is the acceptable amount of time
+ to process the media in a live pipeline before it reaches the sink.
+ This is on top of the systemic latency that is normally reported by
+ the latency query. This defaults to 20ms and should make pipelines
+ such as v4l2src ! xvimagesink not claim that all frames are late in
+ the QoS events. Ideally, this should replace the "max-lateness"
+ property for most applications.
+
+- RTCP Extended Reports (XR) parsing according to RFC 3611:
+ Loss/Duplicate RLE, Packet Receipt Times, Receiver Reference Time,
+ Delay since the last Receiver (DLRR), Statistics Summary, and VoIP
+ Metrics reports. This only provides the ability to parse such
+ packets, generation of XR packets is not supported yet and XR
+ packets are not automatically parsed by rtpbin / rtpsession but must
+ be actively handled by the application.
+
+- a new mode for interlaced video was added where each buffer carries
+ a single field of interlaced video, with buffer flags indicating
+ whether the field is the top field or bottom field. Top and bottom
+ fields are expected to alternate in this mode. Caps for this
+ interlace mode must also carry a format:Interlaced caps feature to
+ ensure backwards compatibility.
+
+- The video library has gained support for three new raw pixel
+ formats:
+
+ - Y410: packed 4:4:4 YUV, 10 bits per channel
+ - Y210: packed 4:2:2 YUV, 10 bits per channel
+ - NV12_10LE40: fully-packed 10-bit variant of NV12_10LE32,
+ i.e. without the padding bits
+
+- GstRTPSourceMeta is a new meta that can be used to transport
+ information about the origin of depayloaded or decoded RTP buffers,
+ e.g. when mixing audio from multiple sources into a single stream. A
+ new "source-info" property on the RTP depayloader base class
+ determines whether depayloaders should put this meta on outgoing
+ buffers. Similarly, the same property on RTP payloaders determines
+ whether they should use the information from this meta to construct
+ the CSRCs list on outgoing RTP buffers.
+
+- gst_sdp_message_from_text() is a convenience constructor to parse
+ SDPs from a string which is particularly useful for language
+ bindings.
+
+Support for Planar (Non-Interleaved) Raw Audio
+
+Raw audio samples are usually passed around in interleaved form in
+GStreamer, which means that if there are multiple audio channels the
+samples for each channel are interleaved in memory, e.g.
+|LEFT|RIGHT|LEFT|RIGHT|LEFT|RIGHT| for stereo audio. A non-interleaved
+or planar arrangement in memory would look like
+|LEFT|LEFT|LEFT|RIGHT|RIGHT|RIGHT| instead, possibly with
+|LEFT|LEFT|LEFT| and |RIGHT|RIGHT|RIGHT| residing in separate memory
+chunks or separated by some padding.
+
+GStreamer has always had signalling for non-interleaved audio since
+version 1.0, but it was never actually properly implemented in any
+elements. audioconvert would advertise support for it, but wasn’t
+actually able to handle it correctly.
+
+With this release we now have full support for non-interleaved audio as
+well, which means more efficient integration with external APIs that
+handle audio this way, but also more efficient processing of certain
+operations like interleaving multiple 1-channel streams into a
+multi-channel stream which can be done without memory copies now.
+
+New API to support this has been added to the GStreamer Audio support
+library: There is now a new GstAudioMeta which describes how data is
+laid out inside the buffer, and buffers with non-interleaved audio must
+always carry this meta. To access the non-interleaved audio samples you
+must map such buffers with gst_audio_buffer_map() which works much like
+gst_buffer_map() or gst_video_frame_map() in that it will populate a
+little GstAudioBuffer helper structure passed to it with the number of
+samples, the number of planes and pointers to the start of each plane in
+memory. This function can also be used to map interleaved audio buffers
+in which case there will be only one plane of interleaved samples.
+
+Of course support for this has also been implemented in the various
+audio helper and conversion APIs, base classes, and in elements such as
+audioconvert, audioresample, audiotestsrc, audiorate.
+
+Support for Closed Captions and Other Ancillary Data in Video
+
+The video support library has gained support for detecting and
+extracting Ancillary Data from videos as per the SMPTE S291M
+specification, including:
+
+- a VBI (Vertical Blanking Interval) parser that can detect and
+ extract Ancillary Data from Vertical Blanking Interval lines of
+ component signals. This is currently supported for videos in v210
+ and UYVY format.
+
+- a new GstMeta for closed captions: GstVideoCaptionMeta. This
+ supports the two types of closed captions, CEA-608 and CEA-708,
+ along with the four different ways they can be transported (other
+ systems are a superset of those).
+
+- a VBI (Vertical Blanking Interval) encoder for writing ancillary
+ data to the Vertical Blanking Interval lines of component signals.
+
+The new closedcaption plugin in gst-plugins-bad then makes use of all
+this new infrastructure and provides the following elements:
+
+- cccombiner: a closed caption combiner that takes a closed captions
+ stream and another stream and adds the closed captions as
+ GstVideoCaptionMeta to the buffers of the other stream.
+
+- ccextractor: a closed caption extractor which will take
+ GstVideoCaptionMeta from input buffers and output them as a separate
+ closed captions stream.
+
+- ccconverter: a closed caption converter that can convert between
+ different formats
+
+- line21decoder: extract line21 closed captions from SD video streams
+
+- cc708overlay: decodes CEA 608/708 captions and overlays them on
+ video
+
+Additionally, the following elements have also gained Closed Caption
+support:
+
+- qtdemux and qtmux support CEA 608/708 Closed Caption tracks
+
+- mpegvideoparse extracts Closed Captions from MPEG-2 video streams
+
+- decklinkvideosink can output closed captions and decklinkvideosrc
+ can extract closed captions
+
+- playbin and playbin3 learned how to autoplug CEA 608/708 CC overlay
+ elements
+
+- the externally maintained ajavideosrc element for AJA capture cards
+ has support for extracting closed captions
+
+The rsclosedcaption plugin in the Rust plugins collection includes a
+MacCaption (MCC) file parser and encoder.
+
+New Elements
+
+- overlaycomposition: New element that allows applications to draw
+ GstVideoOverlayCompositions on a stream. The element will emit the
+ "draw" signal for each video buffer, and the application then
+ generates an overlay for that frame (or not). This is much more
+ performant than e.g. cairooverlay for many use cases, e.g. because
+ pixel format conversions can be avoided or the blitting of the
+ overlay can be delegated to downstream elements (such as
+ gloverlaycompositor). It’s particularly useful for cases where only
+ a small section of the video frame should be drawn on.
+
+- gloverlaycompositor: New OpenGL-based compositor element that
+ flattens any overlays from GstVideoOverlayCompositionMetas into the
+ video stream. This element is also always part of glimagesink.
+
+- glalpha: New element that adds an alpha channel to a video stream.
+ The values of the alpha channel can either be set to a constant or
+ can be dynamically calculated via chroma keying. It is similar to
+ the existing alpha element but based on OpenGL. Calculations are
+ done in floating point so results may not be identical to the output
+ of the existing alpha element.
+
+- rtpfunnel funnels together RTP streams into a single session. Use
+ cases include multiplexing and bundle. webrtcbin uses it to
+ implement BUNDLE support.
+
+- testsrcbin is a source element that provides an audio and/or video
+ stream and also announces them using the recently-introduced
+ GstStream API. This is useful for testing elements such as playbin3
+ or uridecodebin3 etc.
+
+- New closed caption elements: cccombiner, ccextractor, ccconverter,
+ line21decoder and cc708overlay (see above)
+
+- wpesrc: new source element acting as a Web Browser based on WebKit
+ WPE
+
+- Two new OpenCV-based elements: cameracalibrate and cameraundistort
+ that can communicate to figure out distortion correction parameters
+ for a camera and correct for the distortion.
+
+- New sctp plugin based on usrsctp with sctpenc and sctpdec elements.
+ These elements are used inside webrtcbin for implementing data
+ channels.
+
+New element features and additions
+
+- playbin3, playbin and playsink have gained a new "text-offset"
+ property to adjust the positioning of the selected subtitle stream
+ vis-a-vis the audio and video streams. This uses subtitleoverlay’s
+ new "subtitle-ts-offset" property. GstPlayer has gained matching API
+ for this, namely gst_player_get_text_video_offset().
+
+- playbin3 buffering improvements: in network playback scenarios there
+ may be multiple inputs to decodebin3, and buffering will be done
+ before decodebin3 using queue2 or downloadbuffer elements inside
+ urisourcebin. Since this is before any parsers or demuxers there may
+ not be any bitrate information available for the various streams, so
+ it was difficult to configure the buffering there smartly within
+ global constraints. This was improved now: The queue2 elements
+ inside urisourcebin will now use the new bitrate query to figure out
+ a bitrate estimate for the stream if no bitrate was provided by
+ upstream, and urisourcebin will use the bitrates of the individual
+ queues to distribute the globally-set "buffer-size" budget in bytes
+ to the various queues. urisourcebin also gained "low-watermark" and
+ "high-watermark" properties which will be proxied to the internal
+ queues, as well as a read-only "statistics" property which allows
+ querying of the minimum/maximum/average byte and time levels of the
+ queues inside the urisourcebin in question.
+
+- splitmuxsink has gained a couple of new features:
+
+ - new "async-finalize" mode: This mode is useful for muxers or
+ outputs that can take a long time to finalize a file. Instead of
+ blocking the whole upstream pipeline while the muxer is doing
+ its stuff, we can unlink it and spawn a new muxer + sink
+ combination to continue running normally. This requires us to
+ receive the muxer and sink (if needed) as factories via the new
+ "muxer-factory" and "sink-factory" properties, optionally
+ accompanied by their respective properties structures (set via
+ the new "muxer-properties" and "sink-properties" properties).
+ There are also new "muxer-added" and "sink-added" signals in
+ case custom code has to be called for them to configure them.
+
+ - "split-at-running-time" action signal: When called by the user,
+ this action signal ends the current file (and starts a new one)
+ as soon as the given running time is reached. If called multiple
+ times, running times are queued up and processed in the order
+ they were given.
+
+ - "split-after" action signal to finish outputting the current GOP
+ to the current file and then start a new file as soon as the GOP
+ is finished and a new GOP is opened (unlike the existing
+ "split-now" which immediately finishes the current file and
+ writes the current GOP into the next newly-started file).
+
+ - "reset-muxer" property: when unset, the muxer is reset using
+ flush events instead of setting its state to NULL and back. This
+ means the muxer can keep state across resets, e.g. mpegtsmux
+ will keep the continuity counter continuous across segments as
+ required by hlssink2.
+
+- qtdemux gained PIFF track encryption box support in addition to the
+ already-existing PIFF sample encryption support, and also allows
+ applications to select which encryption system to use via a
+ "drm-preferred-decryption-system-id" context in case there are
+ multiple options.
+
+- qtmux: the "start-gap-threshold" property determines now whether an
+ edit list will be created to account for small gaps or offsets at
+ the beginning of a stream in case the start timestamps of tracks
+ don’t line up perfectly. Previously the threshold was hard-coded to
+ 1% of the (video) frame duration, now it is 0 by default (so edit
+ list will be created even for small differences), but fully
+ configurable.
+
+- rtpjitterbuffer has improved end-of-stream handling
+
+- rtpmp4vpay will be prefered over rtpmp4gpay for MPEG-4 video in
+ autoplugging scenarios now
+
+- rtspsrc now allows applications to send RTSP SET_PARAMETER and
+ GET_PARAMETER requests using action signals.
+
+- rtspsrc has a small (100ms) configurable teardown delay by default
+ to try and make sure an RTSP TEARDOWN request gets sent out when the
+ source element shuts down. This will block the downward PAUSED to
+ READY state change for a short time, but can be disabled where it’s
+ a problem. Some servers only allow a limited number of concurrent
+ clients, so if no proper TEARDOWN is sent new clients may have
+ problems connecting to the server for a while.
+
+- souphttpsrc behaves better with low bitrate streams now. Before it
+ would increase the read block size too quickly which could lead to
+ it not reading any data from the socket for a very long time with
+ low bitrate streams that are output live downstream. This could lead
+ to servers kicking off the client.
+
+- filesink: do internal buffering to avoid performance regression with
+ small writes since we bypass libc buffering by using writev()
+ instead of fwrite()
+
+- identity: add "eos-after" property and fix "error-after" property
+ when the element is reused
+
+- input-selector: lets context queries pass through, so that
+ e.g. upstream OpenGL elements can use contexts and displays
+ advertised by downstream elements
+
+- queue2: avoid ping-pong between 0% and 100% buffering messages if
+ upstream is pushing buffers larger than one of its limits, plus
+ performance optimisations
+
+- opusdec: new "phase-inversion" property to control phase inversion.
+ When enabled, this will slightly increase stereo quality, but
+ produces a stream that when downmixed to mono will suffer audio
+ distortions.
+
+- The x265enc HEVC encoder also exposes a "key-int-max" property to
+ configure the maximum allowed GOP size now.
+
+- decklinkvideosink has seen stability improvements for long-running
+ pipelines (potential crash due to overflow of leaked clock refcount)
+ and clock-slaving improvements when performing flushing seeks
+ (causing stalls in the output timeline), pausing and/or buffering.
+
+- srtpdec, srtpenc: add support for MKIs which allow multiple keys to
+ be used with a single SRTP stream
+
+- The srt Secure Reliable Transport plugin has integrated server and
+ client elements srt{client,server}{src,sink} into one (srtsrc and
+ srtsink), since SRT connection mode can be changed by uri
+ parameters.
+
+- h264parse and h265parse will handle SEI recovery point messages and
+ mark recovery points as keyframes as well (in addition to IDR
+ frames)
+
+- webrtcbin: "add-turn-server" action signal to pass multiple ICE
+ relays (TURN servers).
+
+- The removesilence element has received various new features and
+ properties, such as a "threshold" property, detecting silence only
+ after minimum silence time/buffers, a "silent" property to control
+ bus message notifications as well as a "squash" property.
+
+- AOMedia AV1 decoder gained support for 10/12bit decoding whilst the
+ AV1 encoder supports more image formats and subsamplings now and
+ acquired support for rate control and profile related configuration.
+
+- The Fraunhofer fdkaac plugin can now be built against the 2.0.0
+ version API and has improved multichannel support
+
+- kmssink now supports unpadded 24-bit RGB and can configure mode
+ setting from video info, which enables display of multi-planar
+ formats such as I420 or NV12 with modesetting. It has also gained a
+ number of new properties: The "restore-crtc" property does what it
+ says on the tin and is enabled by default. "plane-properties" and
+ "connector-properties" can be used to pass custom properties to the
+ DRM.
+
+- waylandsink has a "fullscreen" property now.
+
+Plugin and library moves
+
+- The stereo element was moved from -bad into the existing audiofx
+ plugin in -good. If you get duplicate type registration warnings
+ when upgrading, check that you don’t have a stale stereoplugin lying
+ about somewhere.
+
+GstVideoAggregator, compositor, and OpenGL mixer elements moved from -bad to -base
+
+GstVideoAggregator is a new base class for raw video mixers and muxers
+and is based on GstAggregator. It provides defined-latency mixing of raw
+video inputs and ensures that the pipeline won’t stall even if one of
+the input streams stops producing data.
+
+As part of the move to stabilise the API there were some last-minute API
+changes and clean-ups, but those should mostly affect internal elements.
+Most notably, the "ignore-eos" pad property was renamed to
+"repeat-after-eos" and the conversion code was moved to a
+GstVideoAggregatorConvertPad subclass to avoid code duplication, make
+things less awkward for subclasses like the OpenGL-based video mixer,
+and make the API more consistent with the audio aggregator API.
+
+It is used by the compositor element, which is a replacement for
+‘videomixer’ which did not handle live inputs very well. compositor
+should behave much better in that respect and generally behave as one
+would expected in most scenarios.
+
+The compositor element has gained support for per-pad blending mode
+operators (SOURCE, OVER, ADD) which determines what operator to use for
+blending this pad over the previous ones. This can be used to implement
+crossfading and the available operators can be extended in the future as
+needed.
+
+A number of OpenGL-based video mixer elements (glvideomixer, glmixerbin,
+glvideomixerelement, glstereomix, glmosaic) which are built on top of
+GstVideoAggregator have also been moved from -bad to -base now. These
+elements have been merged into the existing OpenGL plugin, so if you get
+duplicate type registration warnings when upgrading, check that you
+don’t have a stale openglmixers plugin lying about somewhere.
+
+Plugin removals
+
+The following plugins have been removed from gst-plugins-bad:
+
+- The experimental daala plugin has been removed, since it’s not so
+ useful now that all effort is focused on AV1 instead, and it had to
+ be enabled explicitly with --enable-experimental anyway.
+
+- The spc plugin has been removed. It has been replaced by the gme
+ plugin.
+
+- The acmmp3dec and acmenc plugins for Windows have been removed. ACM
+ is an ancient legacy API and there was no point in keeping the
+ plugins around for a licensed MP3 decoder now that the MP3 patents
+ have expired and we have a decoder in -good. We also didn’t ship
+ these in our cerbero-built Windows packages, so it’s unlikely that
+ they’ll be missed.
+
+
+Miscellaneous API additions
+
+- GstBitwriter: new generic bit writer API to complement the existing
+ bit reader
+
+- gst_buffer_new_wrapped_bytes() creates a wrap buffer from a GBytes
+
+- gst_caps_set_features_simple() sets a caps feature on all the
+ structures of a GstCaps
+
+- New GST_QUERY_BITRATE query: This allows determining from downstream
+ what the expected bitrate of a stream may be which is useful in
+ queue2 for setting time based limits when upstream does not provide
+ timing information. tsdemux, qtdemux and matroskademux have basic
+ support for this query on their sink pads.
+
+- elements: there is a new “Hardware” class specifier. Elements
+ interacting with hardware devices should specify this classifier in
+ their element factory class metadata. This is useful to advertise as
+ one might need to put such elements into READY state to test if the
+ hardware is present in the system for example.
+
+- protection: Add a new definition for unspecified system protection,
+ GST_PROTECTION_UNSPECIFIED_SYSTEM_ID
+
+- take functions for various mini objects that didn’t have them yet:
+ gst_query_take(), gst_message_take(), gst_tag_list_take(),
+ gst_buffer_list_take(). Unlike the various _replace() functions
+ _take() does not increase the reference count but takes ownership of
+ the mini object passed.
+
+- clear functions for various mini object types and GstObject which
+ unrefs the object or mini object (if non-NULL) and sets the variable
+ pointed to to NULL: gst_clear_structure(), gst_clear_tag_list(),
+ gst_clear_query(), gst_clear_message(), gst_clear_event(),
+ gst_clear_caps(), gst_clear_buffer_list(), gst_clear_buffer(),
+ gst_clear_mini_object(), gst_clear_object()
+
+- miniobject: new API gst_mini_object_add_parent() and
+ gst_mini_object_remove_parent() to set parent pointers on mini
+ objects to ensure correct writability: Every container of
+ miniobjects now needs to store itself as parent in the child object,
+ and remove itself again later. A mini object is then only writable
+ if there is at most one parent, that parent is writable itself, and
+ the reference count of the mini object is 1. GstBuffer (for
+ memories), GstBufferList (for buffers), GstSample (for caps, buffer,
+ bufferlist), and GstVideoOverlayComposition were updated
+ accordingly. Without this it was possible to have e.g. a buffer list
+ with a refcount of 2 used in two places at once that both modify the
+ same buffer with refcount 1 at the same time wrongly thinking it is
+ writable even though it’s really not.
+
+- poll: add API to watch for POLLPRI and stop treating POLLPRI as a
+ read. This is useful to wait for video4linux events which are
+ signalled via POLLPRI.
+
+- sample: new API to update the contents of a GstSample and make it
+ writable: gst_sample_set_buffer(), gst_sample_set_caps(),
+ gst_sample_set_segment(), gst_sample_set_info(), plus
+ gst_sample_is_writable() and gst_sample_make_writable(). This makes
+ it possible to reuse a sample object and avoid unnecessary memory
+ allocations, for example in appsink.
+
+- ClockIDs now keep a weak reference to underlying clock to avoid
+ crashes in basesink in corner cases where a clock goes away while
+ the ClockID is still in use, plus some new API
+ (gst_clock_id_get_clock(), gst_clock_id_uses_clock()) to check the
+ clock a ClockID is linked to.
+
+- The GstCheck unit test library gained a
+ fail_unless_equals_clocktime() convenience macro as well as some new
+ GstHarness API for for proposing meta APIs from the allocation
+ query: gst_harness_add_propose_allocation_meta(). ASSERT_CRITICAL()
+ checks in unit tests are now skipped if GStreamer was compiled with
+ GST_DISABLE_GLIB_CHECKS.
+
+- gst_audio_buffer_truncate() convenience function to truncate a raw
+ audio buffer
+
+
+Miscellaneous performance and memory optimisations
+
+As always there have been many performance and memory usage improvements
+across all components and modules. Some of them (such as dmabuf
+import/export) have already been mentioned elsewhere so won’t be
+repeated here.
+
+The following list is only a small snapshot of some of the more
+interesting optimisations that haven’t been mentioned in other contexts
+yet:
+
+- The GstVideoEncoder and GstVideoDecoder base classes now release the
+ STREAM_LOCK when pushing out buffers, which means (multi-threaded)
+ encoders and decoders can now receive and continue to process input
+ buffers whilst waiting for downstream elements in the pipeline to
+ process the buffer that was pushed out. This increases throughput
+ and reduces processing latency, also and especially for
+ hardware-accelerated encoder/decoder elements.
+
+- GstQueueArray has seen a few API additions
+ (gst_queue_array_peek_nth(), gst_queue_array_set_clear_func(),
+ gst_queue_array_clear()) so that it can be used in other places like
+ GstAdapter instead of a GList, which reduces allocations and
+ improves performance.
+
+- appsink now reuses the sample object in pull_sample() if possible
+
+- rtpsession only starts the RTCP thread when it’s actually needed now
+
+- udpsrc uses a buffer pool now and the GstUdpSrc object structure was
+ optimised for better cache performance
+
+GstPlayer
+
+- API was added to fine-tune the synchronisation offset between
+ subtitles and video
+
+
+Miscellaneous changes
+
+- As a result of moving to newer FFmpeg APIs, encoder and decoder
+ elements exposed by the GStreamer FFmpeg wrapper plugin (gst-libav)
+ may have seen possibly incompatible changes to property names and/or
+ types, and not all properties exposed might be functional. We are
+ still reviewing the new properties and aim to minimise breaking
+ changes at least for the most commonly-used properties, so please
+ report any issues you run into!
+
+OpenGL integration
+
+- The OpenGL mixer elements have been moved from -bad to
+ gst-plugins-base (see above)
+
+- The Mesa GBM backend now supports headless mode
+
+- gloverlaycompositor: New OpenGL-based compositor element that
+ flattens any overlays from GstVideoOverlayCompositionMetas into the
+ video stream.
+
+- glalpha: New element that adds an alpha channel to a video stream.
+ The values of the alpha channel can either be set to a constant or
+ can be dynamically calculated via chroma keying. It is similar to
+ the existing alpha element but based on OpenGL. Calculations are
+ done in floating point so results may not be identical to the output
+ of the existing alpha element.
+
+- glupload: Implement direct dmabuf uploader, the idea being that some
+ GPUs (like the Vivante series) can actually perform the YUV->RGB
+ conversion internally, so no custom conversion shaders are needed.
+ To make use of this feature, we need an additional uploader that can
+ import DMABUF FDs and also directly pass the pixel format, relying
+ on the GPU to do the conversion.
+
+
+Tracing framework and debugging improvements
+
+- There is now a GDB PRETTY PRINTER FOR VARIOUS GSTREAMER TYPES: For
+ GstObject pointers the type and name is added, e.g.
+ 0x5555557e4110 [GstDecodeBin|decodebin0]. For GstMiniObject pointers
+ the object type is added, e.g. 0x7fffe001fc50 [GstBuffer]. For
+ GstClockTime and GstClockTimeDiff the time is also printed in human
+ readable form, e.g. 150116219955 [+0:02:30.116219955].
+
+- GDB EXTENSION WITH TWO CUSTOM GDB COMMANDS gst-dot AND gst-print:
+
+ - gst-dot creates dot files that a very close to what
+ GST_DEBUG_BIN_TO_DOT_FILE() produces, but object properties and
+ buffer contents such as codec-data in caps are not available.
+
+ - gst-print produces high-level information about a GStreamer
+ object. This is currently limited to pads for GstElements and
+ events for the pads. The output may look like this:
+
+ (gdb) gst-print pad.object.parent
+ GstMatroskaDemux (matroskademux0) {
+ SinkPad (sink, pull) {
+ }
+ SrcPad (video_0, push) {
+ events:
+ stream-start:
+ stream-id: 0463ccb080d00b8689bf569a435c4ff84f9ff753545318ae2328ea0763fd0bec/001:1274058367
+ caps: video/x-theora
+ width: 1920
+ height: 800
+ pixel-aspect-ratio: 1/1
+ framerate: 24/1
+ streamheader: < 0x5555557c7d30 [GstBuffer], 0x5555557c7e40 [GstBuffer], 0x7fffe00141d0 [GstBuffer] >
+ segment: time
+ rate: 1
+ tag: global
+ container-format: Matroska
+ }
+ SrcPad (audio_0, push) {
+ events:
+ stream-start:
+ stream-id: 0463ccb080d00b8689bf569a435c4ff84f9ff753545318ae2328ea0763fd0bec/002:1551204875
+ caps: audio/mpeg
+ mpegversion: 4
+ framed: true
+ stream-format: raw
+ codec_data: 0x7fffe0014500 [GstBuffer]
+ level: 2
+ base-profile: lc
+ profile: lc
+ channels: 2
+ rate: 44100
+ segment: time
+ rate: 1
+ tag: global
+ container-format: Matroska
+ tag: stream
+ audio-codec: MPEG-4 AAC audio
+ language-code: en
+ }
+ }
+
+- gst_structure_to_string() now serialises the actual value of
+ pointers when serialising GstStructures instead of claiming they’re
+ NULL. This makes debug logging in various places less confusing,
+ because it’s clear now that structure fields actually hold valid
+ objects. Such object pointer values will never be deserialised
+ however.
+
+
+Tools
+
+- gst-inspect-1.0 has coloured output now and will automatically use a
+ pager if the output does not fit on a page. This only works in a
+ UNIX environment and if the output is not piped, and on Windows 10
+ build 16257 or newer. If you don’t like the colours you can disable
+ them by setting the GST_INSPECT_NO_COLORS=1 environment variable or
+ passing the --no-color command line option.
+
+
+GStreamer RTSP server
+
+- Improved backlog handling when using TCP interleaved for data
+ transport. Before there was a fixed maximum size for backlog
+ messages, which was prone to deadlocks and made it difficult to
+ control memory usage with the watch backlog. The RTSP server now
+ limits queued TCP data messages to one per stream, moving queuing of
+ the data into the pipeline and leaving the RTSP connection
+ responsive to RTSP messages in both directions, preventing all those
+ problems.
+
+- Initial ULP Forward Error Correction support in rtspclientsink and
+ for RECORD mode in the server.
+
+- API to explicitly enable retransmission requests (RTX)
+
+- Lots of multicast-related fixes
+
+- rtsp-auth: Add support for parsing .htdigest files
+
+
+GStreamer VAAPI
+
+- this section will be filled in in due course
+
+
+GStreamer OMX
+
+- Add support of NV16 format to video encoders input.
+
+- Video decoders now handle the ALLOCATION query to tell upstream
+ about the number of buffers they require. Video encoders will also
+ use this query to adjust their number of allocated buffers
+ preventing starvation when using dynamic buffer mode.
+
+- The OMX_PERFORMANCE debug category has been renamed to OMX_API_TRACE
+ and can now be used to track a widder variety of interactions
+ between OMX and GStreamer.
+
+- Video encoders will now detect frame rate only changes and will
+ inform OMX about it rather than doing a full format reset.
+
+- Various Zynq UltraScale+ specific improvements:
+ - Video encoders are now able to import dmabuf from upstream.
+ - Support for HEVC range extension profiles and more AVC profiles.
+ - We can now request video encoders to generate an IDR using the
+ force key unit event.
+
+
+GStreamer Editing Services and NLE
+
+- this section will be filled in in due course
+
+
+GStreamer validate
+
+- this section will be filled in in due course
+
+
+GStreamer Python Bindings
+
+- add binding for gst_pad_set_caps()
+
+- pygobject dependency requirement was bumped to >= 3.8
+
+- new audiotestsrc, audioplot, and mixer plugin examples, and a
+ dynamic pipeline example
+
+
+GStreamer C# Bindings
+
+- bindings for the GstWebRTC library
+
+
+GStreamer Rust Bindings
+
+The GStreamer Rust bindings are now officially part of the GStreamer
+project and are also maintained in the GStreamer GitLab.
+
+The releases will generally not be synchronized with the releases of
+other GStreamer parts due to dependencies on other projects.
+
+Also unlike the other GStreamer libraries, the bindings will not commit
+to full API stability but instead will follow the approach that is
+generally taken by Rust projects, e.g.:
+
+1) 0.12.X will be completely API compatible with all other 0.12.Y
+ versions.
+2) 0.12.X+1 will contain bugfixes and compatible new feature additions.
+3) 0.13.0 will _not_ be backwards compatible with 0.12.X but projects
+ will be able to stay at 0.12.X without any problems as long as they
+ don’t need newer features.
+
+The current stable release is 0.12.2 and the next release series will be
+0.13, probably around March 2019.
+
+At this point the bindings cover most of GStreamer core (except for most
+notably GstAllocator and GstMemory), and most parts of the app, audio,
+base, check, editing-services, gl, net. pbutils, player, rtsp,
+rtsp-server, sdp, video and webrtc libraries.
+
+Also included is support for creating subclasses of the following types
+and writing GStreamer plugins:
+
+- gst::Element
+- gst::Bin and gst::Pipeline
+- gst::URIHandler and gst::ChildProxy
+- gst::Pad, gst::GhostPad
+- gst_base::Aggregator and gst_base::AggregatorPad
+- gst_base::BaseSrc and gst_base::BaseSink
+- gst_base::BaseTransform
+
+Changes to 0.12.X since 0.12.0
+
+Fixed
+
+- PTP clock constructor actually creates a PTP instead of NTP clock
+
+Added
+
+- Bindings for GStreamer Editing Services
+- Bindings for GStreamer Check testing library
+- Bindings for the encoding profile API (encodebin)
+
+- VideoFrame, VideoInfo, AudioInfo, StructureRef implements Send and
+ Sync now
+- VideoFrame has a function to get the raw FFI pointer
+- From impls from the Error/Success enums to the combined enums like
+ FlowReturn
+- Bin-to-dot file functions were added to the Bin trait
+- gst_base::Adapter implements SendUnique now
+- More complete bindings for the gst_video::VideoOverlay interface,
+ especially
+ gst_video::is_video_overlay_prepare_window_handle_message()
+
+Changed
+
+- All references were updated from GitHub to freedesktop.org GitLab
+- Fix various links in the README.md
+- Link to the correct location for the documentation
+- Remove GitLab badge as that only works with gitlab.com currently
+
+Changes in git master for 0.13
+
+Fixed
+
+- gst::tag::Album is the album tag now instead of artist sortname
+
+Added
+
+- Subclassing infrastructure was moved directly into the bindings,
+ making the gst-plugin crate deprecated. This involves many API
+ changes but generally cleans up code and makes it more flexible.
+ Take a look at the gst-plugins-rs crate for various examples.
+
+- Bindings for CapsFeatures and Meta
+- Bindings for
+ ParentBufferMeta,VideoMetaandVideoOverlayCompositionMeta`
+- Bindings for VideoOverlayComposition and VideoOverlayRectangle
+- Bindings for VideoTimeCode
+
+- UniqueFlowCombiner and UniqueAdapter wrappers that make use of the
+ Rust compile-time mutability checks and expose more API in a safe
+ way, and as a side-effect implement Sync and Send now
+
+- More complete bindings for Allocation Query
+- pbutils functions for codec descriptions
+- TagList::iter() for iterating over all tags while getting a single
+ value per tag. The old ::iter_tag_list() function was renamed to
+ ::iter_generic() and still provides access to each value for a tag
+- Bus::iter() and Bus::iter_timed() iterators around the corresponding
+ ::pop\*() functions
+
+- serde serialization of Value can also handle Buffer now
+
+- Extensive comments to all examples with explanations
+- Transmuxing example showing how to use typefind, multiqueue and
+ dynamic pads
+- basic-tutorial-12 was ported and added
+
+Changed
+
+- Rust 1.31 is the minimum supported Rust version now
+- Update to latest gir code generator and glib bindings
+
+- Functions returning e.g. gst::FlowReturn or other “combined” enums
+ were changed to return split enums like
+ Result<gst::FlowSuccess, gst::FlowError> to allow usage of the
+ standard Rust error handling.
+
+- MiniObject subclasses are now newtype wrappers around the underlying
+ GstRc<FooRef> wrapper. This does not change the API in any breaking
+ way for the current usages, but allows MiniObjects to also be
+ implemented in other crates and makes sure rustdoc places the
+ documentation in the right places.
+
+- BinExt extension trait was renamed to GstBinExt to prevent conflicts
+ with gtk::Bin if both are imported
+
+- Buffer::from_slice() can’t possible return None
+
+- Various clippy warnings
+
+
+GStreamer Rust Plugins
+
+Like the GStreamer Rust bindings, the Rust plugins are now officially
+part of the GStreamer project and are also maintained in the GStreamer
+GitLab.
+
+In the 0.3.x versions this contained infrastructure for writing
+GStreamer plugins in Rust, and a set of plugins.
+
+In git master that infrastructure was moved to the GLib and GStreamer
+bindings directly, together with many other improvements that were made
+possible by this, so the gst-plugins-rs repository only contains
+GStreamer elements now.
+
+Elements included are:
+
+- Tutorials plugin: identity, rgb2gray and sinesrc with extensive
+ comments
+
+- rsaudioecho, a port of the audiofx element
+
+- rsfilesrc, rsfilesink
+
+- rsflvdemux, a FLV demuxer. Not feature-equivalent with flvdemux yet
+
+- threadshare plugin: ts-appsrc, ts-proxysrc/sink, ts-queue, ts-udpsrc
+ and ts-tcpclientsrc elements that use a fixed number of threads and
+ share them between instances. For more background about these
+ elements see Sebastian’s talk “When adding more threads adds more
+ problems - Thread-sharing between elements in GStreamer” at the
+ GStreamer Conference 2017.
+
+- rshttpsrc, a HTTP source around the hyper/reqwest Rust libraries.
+ Not feature-equivalent with souphttpsrc yet.
+
+- togglerecord, an element that allows to start/stop recording at any
+ time and keeps all audio/video streams in sync.
+
+- mccparse and mccenc, parsers and encoders for the MCC closed caption
+ file format.
+
+Changes to 0.3.X since 0.3.0
+
+- All references were updated from GitHub to freedesktop.org GitLab
+- Fix various links in the README.md
+- Link to the correct location for the documentation
+
+Changes in git master for 0.4
+
+- togglerecord: Switch to parking_lot crate for mutexes/condition
+ variables for lower overhead
+- Merge threadshare plugin here
+- New closedcaption plugin with mccparse and mccenc elements
+- New identity element for the tutorials plugin
+
+- Register plugins statically in tests instead of relying on the
+ plugin loader to find the shared library in a specific place
+
+- Update to the latest API changes in the GLib and GStreamer bindings
+- Update to the latest versions of all crates
+
+
+Build and Dependencies
+
+- The MESON BUILD SYSTEM BUILD IS NOW FEATURE-COMPLETE (*) and it is
+ now the recommended build system on all platforms and also used by
+ Cerbero to build GStreamer on all platforms. The Autotools build is
+ scheduled to be removed in the next cycle. Developers who currently
+ use gst-uninstalled should move to gst-build. The build option
+ naming has been cleaned up and made consistent and there are now
+ feature options to enable/disable plugins and various other features
+ on a case-by-case basis. (*) with the exception of plugin docs which
+ will be handled differently in future
+
+- Symbol export in libraries is now controlled via explicit exports
+ using symbol visibility or export defines where supported, to ensure
+ consistency across all platforms. This also allows libraries to have
+ exports that vary based on detected platform features and configure
+ options as is the case with the GStreamer OpenGL integration library
+ for example. A few symbols that had been exported by accident in
+ earlier versions may no longer be exported. These symbols will not
+ have had declarations in any public header files then though and
+ would not have been usable.
+
+- The GStreamer FFmpeg wrapper plugin (gst-libav) now depends on
+ FFmpeg 4.x and uses the new FFmpeg 4.x API and stopped relying on
+ ancient API that was removed with the FFmpeg 4.x release. This means
+ that it is no longer possible to build this module against an older
+ system-provided FFmpeg 3.x version. Use the internal FFmpeg 4.x copy
+ instead if you build using autotools, or use gst-libav 1.14.x
+ instead which targets the FFmpeg 3.x API and _should_ work fine in
+ combination with a newer GStreamer. It’s difficult for us to support
+ both old and new FFmpeg APIs at the same time, apologies for any
+ inconvenience caused.
+
+- Hardware-accelerated Nvidia video encoder/decoder plugins nvdec and
+ nvenc can be built against CUDA Toolkit versions 9 and 10.0 now. The
+ dynlink interface has been dropped since it’s deprecated in 10.0.
+
+- The (optional) OpenCV requirement has been bumped to >= 3.0.0 and
+ the plugin can also be built against OpenCV 4.x now.
+
+- New sctp plugin based on usrsctp (for WebRTC data channels)
+
+Cerbero
+
+Cerbero is a meta build system used to build GStreamer plus dependencies
+on platforms where dependencies are not readily available, such as
+Windows, Android, iOS and macOS.
+
+Cerbero has seen a number of improvements:
+
+- Cerbero has been ported to Python 3 and requires Python 3.5 or newer
+ now
+
+- Source tarballs are now protected by checksums in the recipes to
+ guard against download errors and malicious takeover of projects or
+ websites. In addition, downloads are only allowed via secure
+ transports now and plain HTTP, FTP and git:// transports are not
+ allowed anymore.
+
+- There is now a new fetch-bootstrap command which downloads sources
+ required for bootstrapping, with an optional --build-tools-only
+ argument to match the bootstrap --build-tools-only command.
+
+- The bootstrap, build, package and bundle-source commands gained a
+ new --offline switch that ensures that only sources from the cache
+ are used and never downloaded via the network. This is useful in
+ combination with the fetch and fetch-bootstrap commands that acquire
+ sources ahead of time before any build steps are executed. This
+ allows more control over the sources used and when sources are
+ updated, and is particularly useful for build environments that
+ don’t have network access.
+
+- bootstrap --assume-yes will automatically say ‘yes’ to any
+ interactive prompts during the bootstrap stage, such as those from
+ apt-get or yum.
+
+- bootstrap --system-only will only bootstrap the system without build
+ tools.
+
+- Manifest support: The build manifest can be used in continuous
+ integration (CI) systems to fixate the Git revision of certain
+ projects so that all builds of a pipeline are on the same reference.
+ This is used in GStreamer’s gitlab CI for example. It can also be
+ used in order to re-produce a specific build. To set a manifest, you
+ can set manifest = 'my_manifest.xml' in your configuration file, or
+ use the --manifest command line option. The command line option will
+ take precendence over anything specific in the configuration file.
+
+- The new build-deps command can be used to build only the
+ dependencies of a recipe, without the recipe itself.
+
+- new --list-variants command to list available variants
+
+- variants can now be set on the command line via the -v option as a
+ comma-separated list. This overrides any variants set in any
+ configuration files.
+
+- new qt5, intelmsdk and nvidia variants for enabling Qt5 and hardware
+ codec support. See the Enabling Optional Features with Variants
+ section in the Cerbero documentation for more details how to enable
+ and use these variants.
+
+- A new -t / --timestamp command line switch makes commands print
+ timestamps
+
+
+Platform-specific changes and improvements
+
+Android
+
+- toolchain: update compiler to clang and NDKr18. NDK r18 removed the
+ armv5 target and only has Android platforms that target at least
+ armv7 so the armv5 target is not useful anymore.
+
+- The way that GIO modules are named has changed due to upstream GLib
+ natively adding support for loading static GIO modules. This means
+ that any GStreamer application using gnutls for SSL/TLS on the
+ Android or iOS platforms (or any other setup using static libraries)
+ will fail to link looking for the g_io_module_gnutls_load_static()
+ function. The new function name is now
+ g_io_gnutls_load(gpointer data). data can be NULL for a static
+ library. Look at this commit for the necessary change in the
+ examples.
+
+- various build issues on Android have been fixed.
+
+macOS and iOS
+
+- various build issues on iOS have been fixed.
+
+- the minimum required iOS version is now 9.0. The difference in
+ adoption between 8.0 and 9.0 is 0.1% and the bump to 9.0 fixes some
+ build issues.
+
+- The way that GIO modules are named has changed due to upstream GLib
+ natively adding support for loading static GIO modules. This means
+ that any GStreamer application using gnutls for SSL/TLS on the
+ Android or iOS platforms (or any other setup using static libraries)
+ will fail to link looking for the g_io_module_gnutls_load_static()
+ function. The new function name is now
+ g_io_gnutls_load(gpointer data). data can be NULL for a static
+ library. Look at this commit for the necessary change in the
+ examples.
+
+Windows
+
+- The webrtcdsp element is shipped again as part of the Windows binary
+ packages, the build system issue has been resolved.
+
+- ‘Inconsistent DLL linkage’ warnings when building with MSVC have
+ been fixed
+
+- Hardware-accelerated Nvidia video encoder/decoder plugins nvdec and
+ nvenc build on Windows now, also with MSVC and using Meson.
+
+- The ksvideosrc camera capture plugin supports 16-bit grayscale video
+ now
+
+- The wasapisrc audio capture element implements loopback recording
+ from another output device or sink
+
+- wasapisink recover from low buffer levels in shared mode and some
+ exclusive mode fixes
+
+- dshowsrc now implements the GstDeviceMonitor interface
+
+
+Contributors
+
+Aleix Conchillo Flaqué, Alessandro Decina, Alexandru Băluț, Alex Ashley,
+Alexey Chernov, Alicia Boya García, Amit Pandya, Andoni Morales
+Alastruey, Andreas Frisch, Andre McCurdy, Andy Green, Anthony Violo,
+Antoine Jacoutot, Antonio Ospite, Arun Raghavan, Aurelien Jarno,
+Aurélien Zanelli, ayaka, Bananahemic, Bastian Köcher, Branko Subasic,
+Brendan Shanks, Carlos Rafael Giani, Christoph Reiter, Corentin Noël,
+Daeseok Youn, Daniel Drake, Daniel Klamt, Dardo D Kleiner, David Ing,
+David Svensson Fors, Devarsh Thakkar, Dimitrios Katsaros, Edward Hervey,
+Emilio Pozuelo Monfort, Enrique Ocaña González, Ezequiel Garcia, Fabien
+Dessenne, Fabrizio Gennari, Florent Thiéry, Francisco Velazquez,
+Freyr666, Garima Gaur, Gary Bisson, George Kiagiadakis, Georg Lippitsch,
+Georg Ottinger, Geunsik Lim, Göran Jönsson, Guillaume Desmottes, H1Gdev,
+Haihao Xiang, Haihua Hu, Harshad Khedkar, Havard Graff, He Junyan,
+Hoonhee Lee, Hosang Lee, Hyunjun Ko, Ingo Randolf, Iñigo Huguet, James
+Stevenson, Jan Alexander Steffens, Jan Schmidt, Jerome Laheurte, Jimmy
+Ohn, Joakim Johansson, Jochen Henneberg, Johan Bjäreholt, John-Mark
+Bell, John Nikolaides, Jonathan Karlsson, Jonny Lamb, Jordan Petridis,
+Josep Torra, Joshua M. Doe, Jos van Egmond, Juan Navarro, Jun Xie,
+Junyan He, Justin Kim, Kai Kang, Kim Tae Soo, Kirill Marinushkin, Kyrylo
+Polezhaiev, Lars Petter Endresen, Linus Svensson, Louis-Francis
+Ratté-Boulianne, Luis de Bethencourt, Luz Paz, Lyon Wang, Maciej Wolny,
+Marc-André Lureau, Marc Leeman, Marcos Kintschner, Marian Mihailescu,
+Marinus Schraal, Mark Nauwelaerts, Marouen Ghodhbane, Martin Kelly,
+Matej Knopp, Mathieu Duponchelle, Matteo Valdina, Matthew Waters,
+Matthias Fend, memeka, Michael Drake, Michael Gruner, Michael Olbrich,
+Michael Tretter, Miguel Paris, Mike Wey, Mikhail Fludkov, Naveen
+Cherukuri, Nicola Murino, Nicolas Dufresne, Niels De Graef, Nirbheek
+Chauhan, Norbert Wesp, Ognyan Tonchev, Olivier Crête, Omar Akkila,
+Patricia Muscalu, Patrick Radizi, Patrik Nilsson, Paul Kocialkowski, Per
+Forlin, Peter Körner, Peter Seiderer, Petr Kulhavy, Philippe Normand,
+Philippe Renon, Philipp Zabel, Pierre Labastie, Roland Jon, Roman
+Sivriver, Rosen Penev, Russel Winder, Sam Gigliotti, Sean-Der, Sebastian
+Dröge, Seungha Yang, Sjoerd Simons, Snir Sheriber, Song Bing, Soon,
+Thean Siew, Sreerenj Balachandran, Stefan Ringel, Stephane Cerveau,
+Stian Selnes, Suhas Nayak, Takeshi Sato, Thiago Santos, Thibault
+Saunier, Thomas Bluemel, Tianhao Liu, Tim-Philipp Müller, Tomasz
+Andrzejak, Tomislav Tustonić, U. Artie Eoff, Ulf Olsson, Varunkumar
+Allagadapa, Víctor Guzmán, Víctor Manuel Jáquez Leal, Vincenzo Bono,
+Vineeth T M, Vivia Nikolaidou, Wang Fei, wangzq, Whoopie, Wim Taymans,
+Wind Yuan, Wonchul Lee, Xabier Rodriguez Calvar, Xavier Claessens,
+Haihao Xiang, Yacine Bandou, Yeongjin Jeong, Yuji Kuwabara, Zeeshan Ali,
+
+… and many others who have contributed bug reports, translations, sent
+suggestions or helped testing.
+
+
+Bugs fixed in 1.16
+
+- this section will be filled in in due course
+
+More than XXX bugs have been fixed during the development of 1.16.
+
+This list does not include issues that have been cherry-picked into the
+stable 1.16 branch and fixed there as well, all fixes that ended up in
+the 1.16 branch are also included in 1.16.
+
+This list also does not include issues that have been fixed without a
+bug report in bugzilla, so the actual number of fixes is much higher.
+
+
+Stable 1.16 branch
+
+After the 1.16.0 release there will be several 1.16.x bug-fix releases
+which will contain bug fixes which have been deemed suitable for a
+stable branch, but no new features or intrusive changes will be added to
+a bug-fix release usually. The 1.16.x bug-fix releases will be made from
+the git 1.16 branch, which is a stable branch.
+
+1.16.0
+
+1.16.0 is scheduled to be released in March 2019.
+
+
+Known Issues
+
+- possibly breaking/incompatible changes to properties of wrapped
+ FFmpeg decoders and encoders (see above).
+
+- The way that GIO modules are named has changed due to upstream GLib
+ natively adding support for loading static GIO modules. This means
+ that any GStreamer application using gnutls for SSL/TLS on the
+ Android or iOS platforms (or any other setup using static libraries)
+ will fail to link looking for the g_io_module_gnutls_load_static()
+ function. The new function name is now
+ g_io_gnutls_load(gpointer data). See Android/iOS sections above for
+ further details.
+
+
+Schedule for 1.18
+
+Our next major feature release will be 1.18, and 1.17 will be the
+unstable development version leading up to the stable 1.18 release. The
+development of 1.17/1.18 will happen in the git master branch.
+
+The plan for the 1.18 development cycle is yet to be confirmed, but it
+is expected that feature freeze will be around July 2019 followed by
+several 1.17 pre-releases and the new 1.18 stable release in
+August/September.
+
+1.18 will be backwards-compatible to the stable 1.16, 1.14, 1.12, 1.10,
+1.8, 1.6, 1.4, 1.2 and 1.0 release series.
+
+------------------------------------------------------------------------
+
+_These release notes have been prepared by Tim-Philipp Müller with_
+_contributions from Sebastian Dröge and Guillaume Desmottes._
+
+_License: CC BY-SA 4.0_