-This is GStreamer Good Plug-ins 0.10.7, "Red Door Black"
-
-Changes since 0.10.6:
-
- * 8 months of frenetic development
- * Massive RTSP/RTP improvements
- * Fixes in pngdec, gdkpixbufdec, wavpackparse, wavpackdec, smokeenc,
- mulawdec, alwdec, id3demux
- * More Fixes in matroskademux, udpsrc, apedemux, flacenc, avimux,
- dv1394src
- * V4L2 support improved
- * Fixes for OS/X and Windows video/audio sources and sinks
- * Support more formats in QT files
- * Elements moved from gst-plugins-bad: equalizer, lpwsinc (now
- audiowsinclimit), bpwsinc (now audiowsincband), spectrum,
- multifilesrc/sink
- * New audio effects: High/Low/Band-pass filters
- * Many, many other fixes
-
-Bugs fixed since 0.10.6:
-
- * 415627 : [PLUGIN-MOVE] move equalizer to good
- * 463624 : [PATCH] avimux: mismatch in ODML tag numbering
- * 347848 : [id3v2mux] add support for DiscID tags
- * 348085 : [PLUGIN-MOVE] move spectrum to good
- * 351726 : [rtspsrc] Force to send on periodic time new rtsp play co...
- * 358841 : [dv1394src] should have property probe
- * 417420 : [autoaudiosink] add " caps " property to filter sinks by caps
- * 427573 : [osxaudiosrc] segmentation fault
- * 435435 : [alaw] use tables for ALAW (G.711) encoding and decoding
- * 442034 : [avi] add support for subtitle streams (GAB2)
- * 447000 : [id3demux] add support for reading license URL from WCOP tag
- * 448278 : [v4l2src] several issues in cvs
- * 449747 : Function parameter mismatch breaks build on Solaris.
- * 450190 : [PATCH] v4l2src set_capture (parameters) fails
- * 450878 : Missing files in po/POTFILES.in
- * 451249 : Integer type mismatch in gstrtspsrc.c
- * 451388 : [v4l2src] Could not negotiate format
- * 453037 : rtpilbcpayloader encoding-name field mismatch
- * 453417 : [matroskademux] support palette info for VobSub stream
- * 453630 : Rtspsrc invokes GST_ELEMENT_ERROR in eos when streaming o...
- * 455086 : [cairotextoverlay] ignores the size in font-desc property
- * 455808 : rtspsrc deadlocks on pause
- * 457097 : [faad] " setting caps on pad failed " with BMJ113 mov
- * 461600 : Some files missing from POTFILES.in
- * 464475 : the mode property if the caps of rtpilbc*pay should be of...
- * 464800 : [audiofx] New elements: chebyshev low/highpass and bandpa...
- * 465040 : [RTP] h263ppay should allow h263 caps on sink
- * 465774 : qt video does not play, but can be seeked
- * 467214 : Could not get buffers from device '/dev/video0'
- * 467666 : [ELEMENT-MOVE] Move lpwsinc and bpwsinc to gst-plugins-good
- * 470502 : Ricoh Webcam not working
- * 471364 : [gconf] Internal GStreamer error: state change failed on ...
- * 471823 : [v4l2src] Fails to negotiate capabilities with Macbook Pr...
- * 473670 : [qtdemux] invalid utf8 in tags causes warnings
- * 474616 : [rtph263pay] RTP headers are wrong for win32
- * 475424 : [v4l2] uvcvideo Macbook iSight webcam not working (regres...
- * 477199 : Linux-UVC webcams not working
- * 477456 : [v4l2src] my webcam is not working anymore
- * 479960 : qtdemux is missing fourcc for mpeg2 HDV streams (HDV2. HDV3)
- * 480557 : [v4l2src] Nasty segfault (with broken driver?)
- * 482495 : avimux does not accept audio/aac
- * 484998 : [mulawdec] not-negotiated error playing a sun audio (au) ...
- * 485828 : [v4l2src] Fixates to lowest resolution on new kernels
- * 487488 : udpsrc errors
- * 487563 : Payloader for H264 not marking packets
- * 488112 : [id3demux] extract ID3v2 WOAF frame
- * 488844 : Free GstBuffer with gst_buffer_unref() in gst_rtp_mp4v_pa...
- * 488879 : [videomixer] implement GstChildProxy interface
- * 489940 : [gdkpixbufdec] gst_value_set_fraction: assertion 'denomin...
- * 490034 : [v4l2src] " could not get buffers from device "
- * 490283 : [multifilsrc] move to -good
- * 491323 : Extend RTP MPEG-TS payloader to handle large RTP payload ...
- * 492388 : Build issues on Windows/MSVC
- * 496752 : [rtspsrc] leaks memory
- * 496773 : [rtspsrc] leaks SDP messages
- * 496983 : [PATCH] rtspsrc leaks pads
- * 497007 : [PATCH] rtspsrc leaks RTSP messages in SETUP
- * 497017 : [PATCH] H.264 RTP depayloader leaks memory in setcaps
- * 497292 : gsttextoverlay compile problem in MSVC
- * 497293 : gsttimeoverlay compile error in win32
- * 498181 : Recent change to gst-plugins-base breaks gst-plugins-good
- * 498297 : [v4l2src] put real supported resolutions in caps instead ...
- * 498395 : [pngenc] leaks buffer data memory
- * 498715 : GStreamer fails to decode MPEG1 Video RTP streams when gs...
- * 499178 : unexpected not negotiated and other weird beharviours
- * 499239 : gstrtpmp4adepay
- * 499383 : [rtph263depay] Doesn't read I-frame information from rtp ...
- * 500403 : [PATCH] videobox element doesn't init liboil
- * 501775 : [wavparse] regression in CVS: seeking is broken
- * 502655 : [avimux] creates corrupt avi files (wrong stream numbering)
- * 502814 : Adds profile (hex string), sps and pps (base64 encoded) i...
- * 502966 : GStreamer segfaults when videobox is introduced into a sp...
- * 503023 : Let multipartdemux work in live mode.
- * 504018 : flacdec emits newsegment before being able to respond to ...
- * 504081 : [matroskamux] support for live sources (calculate duratio...
- * 504895 : Build fails: gst/check/gstcheck.h: No such file or direc...
- * 505745 : [avidemux] warning: 'res' may be used uninitialized in th...
- * 506025 : RTSPsrc doesn't handle redirects.
- * 506715 : [PATCH] [flacdec] PICTURE blocks should be retrieved as m...
- * 507642 : v4l2src Crash - Size 4156 is not a multiple of unit size ...
- * 508644 : The clockrate for theora payloading should always be 90000
- * 509298 : Compilation errors in gstavisubtitle.c and gstrtspsrc.c w...
- * 509301 : [flacdec] CVS does not compile against flac-1.1.2
- * 509531 : Make the auto-join/auto-leaving of a multicast group opti...
- * 510505 : changes to make v4l plugin build on solaris
- * 510592 : Race condition in WAVE parser
- * 513628 : [flacdec] Decoding of PICTURE blocks is always disabled b...
- * 514397 : matroskademux does not mark delta units
- * 514573 : multipartmux memory leak
- * 514889 : Bogus output from RTP data test
- * 514965 : Possible buffer leaks if last_write_result != GST_FLOW_OK...
- * 515457 : [audiofx] Element long descriptions are broken
- * 515562 : [avidemux] Seeking broken in 0.10.6.2
- * 515697 : [multifile] Several memory leaks exposed by unit test
- * 515701 : [smoke] Several valgrind warnings exposed by unit test
- * 515703 : [avisubtitle] Several valgrind warnings with unit test (g...
- * 515704 : [gconf] Memory leak when creation of pipeline fails
- * 515706 : [ximagesrc] Memory leak when XDamage support is available
- * 515905 : gst-error.m4 needs no%E_MACRO_REDEFINED on Solaris
- * 515979 : [efence] Doesn't chain up finalize properly
- * 515980 : [directdraw] Does not chain up finalize functions properly
- * 515984 : [v4l] Does not chain up finalize functions properly
- * 515985 : [ximagesrc] regression when using damage
- * 516371 : does not build without xml usage
- * 516524 : uninitialized variables
- * 517386 : [alpha] Doesn't link with libgstbase
- * 508291 : [id3demux] must check if language code in id3v2 COMM fram...
- * 491811 : Rename audiochebyshevfreq* to audiocheb*
- * 447961 : [rtsp] do not store a reference to the result from inet_n...
- * 475784 : rtspsrc has race condition between loop_interleaved() and...
- * 478244 : rtpamrdepay make no playable stream
- * 480707 : Memory leak : gstrtpmp4vpay : config buffer is leaked
- * 502187 : flacdec doesn't send newsegment from the streaming thread
- * 509329 : [rtspsrc] use atoll when parsing rtptime
- * 512544 : preallocate output buffer in pngenc
-
-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
+January/February 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: Monday 14 January 2019, 13:00 UTC (log)_
+
+
+Introduction
+
+The GStreamer team is proud to announce a new major feature release in
+the stable 1.x API series of your favourite cross-platform multimedia
+framework!
+
+As always, this release is again packed with 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
+
+- Spport 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 “v4lsrc ! xvimagesink” not claim that all frames are late in
+ the QoS events. Ideally, this should replace max_lateness 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.
+
+- 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, but it
+was never actually properly implemented in any elements. audioconvert
+would advertise support for it, but wasn’t actually able to handle it.
+
+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 (Video 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 (Video 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 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.
+
+- 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
+ who 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
+
+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 also 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 unset
+ where it’s a problem. Some servers only allow a limited number of
+ concurren clients, so if no proper TEARDOWN is sent 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()
+
+- 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"1 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 gststereo plugin
+ 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][aggregator]. 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.
+
+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 gstopenglmixers 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 them
+ around for a licensed mp3 decoder now that 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
+
+- 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), andGstVideoOverlayComposition`
+ 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 different 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. 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-colors 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 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)
+
+
+Platform-specific changes and improvements
+
+Android
+
+- 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.
+
+macOS and iOS
+
+- macOS binaries should be fully relocatable now
+
+- 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 around January/February 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._
+
+_License: CC BY-SA 4.0_
projects / platform / upstream / gst-plugins-good.git / blobdiff