-This is GStreamer Good Plug-ins 0.10.15, "I've been up all night"
-
-Changes since 0.10.14:
-
- * Some fixes for seeking in wav and FLAC files
- * Faster seeking in Matroska and AVI files
- * RTSP and RTP improvements
- * directdrawsink moved to Bad
- * y4menc and flvmux/flvdemux moved from Bad
- * deinterlace2 moved from Bad, replacing deinterlace
- * Many bug fixes and improvements
- * Pulseaudio sink completely overhauled
-
-Bugs fixed since 0.10.14:
-
- * 572551 : mpeg4videoparse fails to extract codec_data from this net...
- * 577318 : rtspsrc appears to be leaking memory
- * 576286 : [videomixer] Should accept/output non-alpha streams
- * 581333 : rtspsrc: request_pt_map in gstrtspsrc.c fails to return c...
- * 478092 : gstid3v2mux.cc:deprecated conversion from string constant...
- * 486915 : [videomixer] segmentation fault on gst-inspect
- * 509311 : [rtph263pay] rtph263pay does not follow rfc2190
- * 516031 : flac within ogg container can't be played on Jokosher
- * 537537 : [pulse] Latency issues
- * 537609 : RTSP - rtspsrc module support for Scale header
- * 552650 : [rtspsrc] (partially) fails SETUP with MS RTSP servers
- * 562168 : Good plugins' configury overloads --disable-debug
- * 563574 : v4l2src should capture in non-blocking mode
- * 567140 : G726 Packetizer issue for 24kbps & 40 kbps datarate
- * 567857 : [udpsrc] loop on gst_poll_wait when POLLERR because of icmp
- * 570781 : [alawdec] spews ERROR debug messages on shutdown/seek/not...
- * 571153 : [pulsemixer] compiler warnings (on ARM)
- * 571321 : gconfvideo{src,sink} don't disconnect gconf notifications
- * 572256 : gst/avi/gstavidemux.c: Alignment trap in gst_avi_demux_pa...
- * 572358 : law encoders _getcaps ignore rate/channel if not both are...
- * 572413 : [jpegenc] crashes if no input format has been set
- * 573173 : Added Quicktime HTTP tunneling to the RTSP src element
- * 573342 : Unconditioned EAFNOSUPPORT in gstudpnetutils.c
- * 573343 : Type mismatches in gstdirectdrawsink.c
- * 573721 : [PLUGIN-MOVE] move directdrawsink back to -bad
- * 573737 : [PLUGIN-MOVE] Move FLV to -good
- * 574270 : [rtspsrc] Range request is wrong (should say: npt=now-)
- * 574275 : flacdec ! appsink with a seek seems to drop the first buffer
- * 577468 : [id3demux] Frames not extracted if tag is unsynchronised
- * 577609 : [id3v2mux] write RVA2 frames for peak/gain volume data
- * 577671 : [rtspsrc] deadlock on shutdown (locking order problem?)
- * 578052 : gstavidemux: support seeking and duration query in defaul...
- * 578135 : [qtdemux] missing 3gpp Asset metadata handling
- * 578310 : [matroskamux] - Should suppport speex
- * 579070 : [sunaudio] fix compiler warnings
- * 579422 : flacdec can block allocating before it sent a new-segment
- * 579808 : [jpegdec] Doesn't support additional 0xff before end marker
- * 580746 : [qtdemux] 3GPP classification entity byte order reversed
- * 580783 : [PLUGIN-MOVE] Move y4menc to -good
- * 580851 : rtspsrc: various; sanity of ranges, setting of base_time ...
- * 580880 : gstrtpjpegpay is not functioning properly; rtp jpeg paylo...
- * 581329 : rtspsrc: NAT dummy packets not being sent
- * 581568 : ability for replaygain plugin to post level messages
- * 581806 : [souphttpsrc] Should allow overriding the referer
- * 581884 : [PLUGIN-MOVE] Move deinterlace2 to gst-plugins-good
- * 582252 : rganalysis test broken by recent commit
- * 582281 : [rtp] Forgets to link to $(LIBM)
- * 582387 : [avidemux] Seeking regression
- * 582420 : flacdec unit test broken on PPC
- * 582661 : [deinterlace] Fix latency query to return unbound max lat...
- * 582715 : gcc warnings about unitialized
- * 582753 : flacdec check fails
- * 582794 : rganalysis unit test fails with git core
- * 568278 : [qtdemux] add support for vob subtitle streams
- * 569611 : GStreamer videobox element draws thin green lines on edge...
- * 571294 : [matroskamux] Should ignore framerate of 0/1
- * 574169 : avidemux/theoradec don't work well together
- * 575234 : Network interface selection for multicasting with the udp...
- * 576729 : [rtspsrc] perform EOS handling earlier
- * 578257 : Image problems using rtpjpeg(de)pay
- * 579069 : rtp h263pay build fixes
- * 580554 : PATCH: qtdemux: fix demuxing of m4v streams with ac-3 audio
- * 581432 : [multipartdemux] source pads are leaked
- * 581444 : [multipartdemux] free memory read of buffer timestamp
- * 582218 : Uninitialized variable may be used in gstavidemux.c
- * 575937 : udp/gstudpnetutils.c: ip_mreqn unavailable on Solaris (an...
-
-
-Changes since 0.10.13:
-
- * Add autodetect source elements
- * Improvements in RTP payload/depayload and RTSP
- * Support float input in wav, and require depth == width
- * Support inverted RGB video in avi
- * Compilation fixes and smarter format selection in V4L2 support
- * Use libv4l when available
- * Don't install static plugin libraries any more
- * Matroska muxing: Add Dirac, fix AAC
- * Improve qtdemux segment handling
- * Add presets to equalizer
- * OS/X video and audio output improvements
- * Rework Pulseaudio audio output
- * Support basic and digest auth in souphttpsrc
- * Use libsoup-gnome instead of libsoup when available
- * DV demuxer fixes
- * New IIR and FIR base classes and echo filter in audiofx plugin
- * Improved spectrum analysis plugin
- * 8 bit greyscale support in v4l2src and videocrop
- * New aspectratiocrop element
- * Many other bug fixes and improvements
-
-Bugs fixed since 0.10.13:
-
- * 561502 : Not-negociated error in udpsrc
- * 522183 : add preset suport to gstequalizer
- * 523813 : [autodetect] add autovideosrc and autoaudiosrc
- * 557709 : goom should have license headers in all source files
- * 560155 : segfault when trying to set tag to NULL with avimux
- * 527951 : [new element] aspectratiocrop
- * 529379 : RTSP seeks when pipeline is in STATE_PAUSED
- * 532409 : RTP payload/depayloader for G.729
- * 537539 : [pulse] Rhythmbox starts using 100% of CPU time when puls...
- * 537540 : [pulse] Causes a deadlock when the pulseserver dies
- * 537543 : [pulse] sink/src should have " volume " property (was GstMi...
- * 545033 : [v4l2src] Add support for camera specific formats
- * 552140 : [soup] use libsoup-gnome for proxy configuration if avail...
- * 556019 : Add the ability to make souphttpsrc act as a live source
- * 556484 : Implement getcaps in rtpL16pay
- * 556641 : rtpL16depay doesn't support multichannel audio
- * 556802 : Misspelled messages
- * 556955 : [videocrop] fails to renegotiate
- * 556986 : pulsesink deadlocks when raising an error
- * 557260 : [equalizer] Calculates coefficients for every buffer
- * 557293 : Argument type mismatch on Windows
- * 557294 : cross mingw32 gcc: getaddrinfo is not in ws2_32.dll befor...
- * 557710 : Memory leak related to matroskamux's request pads
- * 558427 : rtpmpapay/depay are too wide
- * 558638 : Compiler error on v4l2
- * 558711 : gst-plugins-good-0.10.11 compile fails
- * 559288 : [qtdemux] deadly embrace!
- * 559545 : rtspsrc could not receive data from RTSP server because o...
- * 559547 : Seek bugs when playing rtsp video in m.youtube.com
- * 560641 : rtp mp4 payloader doesn't flush the last frame upon EOS
- * 560756 : New Plugin: JPEG RTP Payloader
- * 561625 : Rtspsrc set segment's last_stop incorrectly
- * 561775 : souphttpsrc patch to support basic and digest authentication
- * 561802 : DLNA specific HTTP header missing
- * 561990 : qtdemux build fails on mingw due to missing definition of...
- * 562434 : [wavenc] Add support for a-law and mu-law encoded wav files
- * 562572 : multiudpsink handles errors in _render() too aggressively
- * 563414 : Fix for SunAudioMixer plugin
- * 563504 : add libv4l2 support to v4l2 gst plugin - required for mos...
- * 563509 : [rtp_data_test] Complains about negotiation errors with m...
- * 563510 : [videocrop] 1x1 unit test fails
- * 564437 : rtpjpegdepay was unable to handle frame dimensions greate...
- * 564948 : Fix and clean up osxaudio
- * 565850 : mov files can contain xvid, fmp4 and ac3 tracks
- * 566616 : v4l2src: b/w support
- * 566843 : [pulse] Terminating an audio stream hangs the application
- * 567577 : [audiofx] Add generic IIR/FIR audio filter elements
- * 567642 : spectrum element has undocumented arbitrary limitation on...
- * 567746 : fix for 547227 created a ref loop
- * 567794 : [PATCH] Rework pulse plugin
- * 567800 : Mp4 created by Avidemux don't play.
- * 567853 : osxaudiosrc does not build on 10.4
- * 567874 : [audiofx] Echo/Reverb filter
- * 567955 : [spectrum] Need compensation for rounding errors
- * 567992 : [videocrop] wrong mouse navigation commands while cropping
- * 568395 : [audiofx] Rename audioreverb to audioecho
- * 568780 : elements/videocrop check fails
- * 568809 : Patch needed for building gst-plugins-good 0.10.13 on Sol...
- * 569820 : In some cases v4l2src element does not return correct norm
- * 570343 : equalizer bands != specified parameters
- * 570435 : Read Title element of Matroska files
- * 571038 : [osxvideosink] build fail
- * 571150 : aspectratiocrop : Memory leak
- * 571153 : [pulsemixer] compiler warnings (on ARM)
- * 571204 : [pulsesink] PA assertion failure triggers abort
- * 570581 : osxvideosink: non-embedded mode is badly broken and shoul...
- * 341752 : [rtspsrc] 500 Internal Server Error response with tagessc...
- * 420658 : gconfaudiosink is not behaving normally
- * 558554 : [avidemux] avidemux not recognizing certain uncompressed ...
- * 561580 : GStreamer does not play a specific WAV file
- * 563056 : " rtpjpegdepay ! jpegdec " doesnt works
- * 565441 : Smoke encoder doesn't set caps correctly
- * 567952 : videocrop: add b/w support
-
-Changes since 0.10.12:
-
- * Fix bad autopoint substitution in the po subdir
-
-Changes since 0.10.11:
-
- * Fix for security advisory TKADV2009-0xx
-
-Changes since 0.10.10:
-
- * HDV capture support
- * Port flactag element to 0.10
- * Support FLAC in alternate bit-depths and more samplerates
- * Matroska muxing improvements
- * Support Google RTSP variant
- * Many other bug-fixes and improvements
-
-Bugs fixed since 0.10.10:
-
- * 545433 : [wavparse] support tempo tag
- * 311586 : Matroska mux/demux don't support flac
- * 350830 : [raw1394] Add HDV support
- * 413841 : [flactag] needs porting to 0.10.x
- * 536067 : [matroska] Add support for real audio/video
- * 537361 : videomixer segment handler fail
- * 543101 : The udpsrc/udpsink/multiudpsink/dynudpsink elements may l...
- * 544956 : [videobox] segfault when changing properties at running p...
- * 545463 : autodetect plugin links to gconf, unnecessarily
- * 545710 : [rtspsrc] : create rtp/rtcp socket pair + same port for d...
- * 546465 : can't play m.youtube.com rtsp streams
- * 547075 : [speex] Encoder doesn't handle non-zero start timestamps ...
- * 547217 : Implement GstPropertyProbe interface on pulsesrc for dete...
- * 547227 : Implement GstPropertyProbe interface on pulsesink for det...
- * 547518 : pulsemixer probe fails after device property is set
- * 547519 : Add device-name property to pulsesrc
- * 548530 : wavenc allows audio depth of 25
- * 548831 : matroska demuxer setting incorrect timestamps
- * 549073 : buffer_alloc function from new basetransform sometimes fa...
- * 549090 : v4l2src has no property named norm but notifies it
- * 549551 : conversion of gst structure names to mime types in multip...
- * 549784 : Memory Corruption Probing v4l2src with Video4Linux1 device
- * 550015 : build failure: gstcmmlparser.c:115: error: format not a s...
- * 550791 : [flacenc] files are too big, mid-side-stereo=true should ...
- * 551048 : rtsp implementation sends PAUSE even when server doesn't ...
- * 551570 : plugin ximagesrc is memory leaking...
- * 551584 : Crash when modifying the videobox " top " and " left " live w...
- * 552213 : [qtdemux] map tiff fourcc to image/tiff caps
- * 553191 : udpsrc: bind() fails on OS X
- * 554771 : totem spins CPU 100% while allocating (infinitely) becaus...
- * 556010 : AVI muxer segfault
- * 556381 : [icydemux] test fails with latest core
- * 556424 : [avidemux] segmentation fault when creating indexes
- * 557085 : [videobox] support dynamically changing properties
- * 557610 : udpsrc fails to bind() on Mac OS X (IPv4 vs. IPv6 problem)
- * 547842 : Vorbis RTP RFC 5215 is out
- * 550288 : Sound not played - a quicktime file
-
-Changes since 0.10.9:
-
- * Move the libcdio cddasrc element to -ugly, because of the GPL license
-
-Bugs fixed since 0.10.9:
-
- * 413705 : [PLUGINS MOVE] gst-plugins-good contails GPL'ed libcdio p...
-
-Changes since 0.10.8:
-
- * Replaygain elements moved from Bad
- * Interleave/Deinterleave elements moved from Bad
- * Pulseaudio plugin http://pulseaudio.org is now integrated
- * New simple Karaoke audio effect plugin
- * Improvements in v4l2src
- * Multi-channel FLAC file fixes
- * AVI and Quicktime reverse playback support
- * AVI and Matroska muxing improvements
- * New element for rendering SMPTE transitions into alpha channels
- * Many improvements in the Win32 directdraw elements
- * Error out cleanly for encrypted streams
- * RTP/UDP handling improvements
- * RTSP digest authentatication implemented.
- * New RTP Pay/Depay-loaders for Speex, G.729, DV & raw video/audio
- * Error concealment for Speex
- * Many other bug-fixes and enhancements
-
-Bugs fixed since 0.10.8:
-
- * 541787 : incorrect timestamp
- * 413705 : [PLUGINS MOVE] gst-plugins-good contails GPL'ed libcdio p...
- * 345393 : Missing documentation
- * 400679 : [PLUGIN-MOVE] pulse-gst to -good
- * 422917 : G729 payloader/depayloader
- * 429322 : [matroska-demux] hangs if file starts at non-zero timestamp
- * 465146 : Add more complete negotiation to gstrtpbasepayload
- * 469917 : multiudpsink IPv6 and diffserv TOS/TC markup
- * 499318 : [osxvideosink] warnings on build (ppc, 64bit, 10.4)
- * 503288 : rtph264pay does not split multiple NAL units in incoming ...
- * 511489 : avimux muxed audio and video out of sync
- * 512345 : gstalpha: chroma-key should not set alpha to 0 if brightn...
- * 515962 : [udp] add ipv6 support to multiudpsink and udpsrc, add mu...
- * 516509 : gstid3v2mux.cc build failure on Solaris, CFLAGS not valid...
- * 519301 : avimux fails when going from NULL to playing a second time
- * 519460 : 8 bytes missing in AVI streamheader
- * 520092 : v4l2src proposed patches
- * 520885 : directdrawsink has some issues
- * 527865 : [PATCH] [matroskademux] uses wrong colourspace for palett...
- * 529454 : [udpsrc] generates false error on win32 due to ICMP dest-...
- * 529692 : [goom] fails to build on Windows with mingw32
- * 529707 : [rtp] fails to build on Windows with mingw32
- * 530886 : Support more mpeg2 variants in .mov files
- * 531532 : rtspsrc leaks filedescriptors when gst_rtspsrc_open fails
- * 531672 : [qtdemux] does not handle files with edit lists correctly...
- * 532295 : gconf setting default audio sink to video default visualizer
- * 532393 : [dvdec] Shouldn't be GST_RANK_PRIMARY
- * 532409 : RTP payload/depayloader for G.729
- * 532423 : RTP payload/depayloader for DV
- * 533264 : Make v4l2 compile without X
- * 533619 : udpsink dies when Linux fails (for no reason) sendto() wi...
- * 535121 : [videomixer] proper seek/segment handling
- * 535300 : [avidemux] missing support for reverse playback.
- * 535935 : [qtdemux] Missing metadata
- * 536228 : [PLUGIN-MOVE] Move interleave/deinterleave to gst-plugins...
- * 536317 : [v4l2src] Doesn't compile with latest git kernel
- * 536646 : v4l2src doesn't pick the optimal resolution
- * 536831 : v4l2src == non-experimental
- * 536903 : udpsrc now assumes one is using multicast
- * 537021 : [cdio] does not extract album title from CD-TEXT
- * 537031 : Enhancement to SunAudio mixer plugin
- * 537361 : videomixer segment handler fail
- * 537377 : Gstreamer avidemux fails on avi files with non-AV data st...
- * 537622 : [matroskademux] Doesn't export covers (file attachments)
- * 537675 : rtp vorbis/theora payloader static caps don't specify del...
- * 537832 : rtsp is broken
- * 539372 : udpsrc do not set port properly on anonymous binding
- * 539548 : [udpsrc] Fails to bind with permission denied when trying...
- * 540067 : souphttpsrc does not support http authentication
- * 540300 : [souphttpsrc] souphttpsrc does not seek ahead in http stream
- * 540940 : Quicktime demuxer does not handle dirac in QT files
- * 541081 : [v4l2src] Doesn't compile with on old kernel
- * 541384 : cross mingw32 gcc: making goom compile
- * 541412 : cross mingw32 gcc: making libgstrtp plugin compile
- * 541650 : udpsrc fails to parse URI with IPv6 address
- * 541956 : v4l2src only requests interlaced video
- * 542410 : qtdemux not recognizing pcm 8bit streams correctly
- * 543054 : [osxvideosink] minor build issue
- * 543255 : -Wno-attributes does not exist in every gcc version
- * 543259 : warning in gstv4l2src.c due to an uninitialized variable
- * 543300 : [qtdemux] [PATCH] wrong depth in RLE video caps and wrong...
- * 544509 : [qtdemux] Regression in segment.stop
- * 539482 : [avidemux] Doesn't parse vprp ODML header
- * 544433 : Some -good plugins aren't buildable with MSVC
- * 536994 : Improve v4l2src negotiation a bit more
- * 329198 : GStreamer-CRITICAL: gst_pad_activate_pull: assertion `old...
- * 532065 : Implement RTSP Digest authentication
- * 533287 : Mapping of mime types to gst names in multipartdemux
- * 538891 : G.726 pay/depayloader is missing
-
-Changes since 0.10.7:
-
- * The libsoup based http source moved from the -bad module and
- was raised to primary rank
- * GOOM visualisation plugin updated to Goom2k4
- * Fixes in UDP, RTSP, matroska, QT/mov, AVI, v4l2, OS/X, SunAudio,
- mulaw and wavpack
- * New gdkpixbuf output for snapshotting
- * Various other bug-fixes
-
-Bugs fixed since 0.10.7:
-
- * 516509 : gstid3v2mux.cc build failure on Solaris, CFLAGS not valid...
- * 515978 : [ladspa] Should be removed from -good until it's ready to...
- * 516649 : [v4l2src] tries to VIDIOC_S_PARM without checking capabil...
- * 517237 : [rtspsrc] Better error when bandwidth is too small
- * 517933 : [tests] trivial C89 fix
- * 518188 : [spectrum] " message " property documentation unclear
- * 518213 : [goom2k4] artefacts on x86-64
- * 518564 : goom dies with illegal instruction
- * 519088 : gst_mulawdec_chain() may call gst_util_uint64_scale_int()...
- * 519417 : Unable to modify DEFAULT_VIDEOSRC in configure
- * 520073 : [goom] (goom2k4) shows mostly black frames
- * 520764 : double buffer unref in speexenc error handling
- * 520880 : audiofx doesn't build on Windows/MSVC
- * 520888 : udpsrc has some issues on Windows
- * 521102 : h.263+ rtp depayloader crashes on short payloads
- * 521875 : [osssrc] recording corrupts upon _get_caps
- * 522278 : Problem with " movq " at file gst-plugins-good/gst/goom/xmmx.c
- * 522767 : [goom] does not build on ppc
- * 523124 : [PLUGIN-MOVE] move souphttpsrc from -bad to -good
- * 523134 : osxvideosink does not stop when you control-C
- * 524593 : [sunaudio] fix mixer track range
- * 525359 : [alaw] [mulaw] Use gst_pad_alloc_buffer_and_set_caps to c...
- * 525833 : [halaudiosrc] unable to get mic USB working with gnome-so...
- * 525860 : [sunaudio] copyright fixes
- * 525946 : [gdkpixbuf] add gdkpixbufsink element
- * 526557 : [matroskademux] seeking regression in CVS
- * 527848 : [souphttpsrc] Give souphttpsrc PRIMARY rank
- * 527984 : Doesn't dist all m4 files needed by configure.ac
- * 527999 : [goom] Doesn't dist goomsl.h
- * 528143 : [goom] fails to build on 32-bit ppc
- * 528615 : Hide goom2k4 logo
- * 529268 : [goom] States unit test exposes some memory leaks
-
-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 1.16.0 was originally released on 19 April 2019.
+
+See https://gstreamer.freedesktop.org/releases/1.16/ for the latest
+version of this document.
+
+_Last updated: Friday 19 April 2019, 00:00 UTC (log)_
+
+
+Introduction
+
+The GStreamer team is proud to announce a new major feature release in
+the stable 1.x API series of your favourite cross-platform multimedia
+framework!
+
+As always, this release is again packed with many new features, bug
+fixes and other improvements.
+
+
+Highlights
+
+- GStreamer WebRTC stack gained support for data channels for
+ peer-to-peer communication based on SCTP, BUNDLE support, as well as
+ support for multiple TURN servers.
+
+- AV1 video codec support for Matroska and QuickTime/MP4 containers
+ and more configuration options and supported input formats for the
+ AOMedia AV1 encoder
+
+- Support for Closed Captions and other Ancillary Data in video
+
+- Support for planar (non-interleaved) raw audio
+
+- GstVideoAggregator, compositor and OpenGL mixer elements are now in
+ -base
+
+- New alternate fields interlace mode where each buffer carries a
+ single field
+
+- WebM and Matroska ContentEncryption support in the Matroska demuxer
+
+- new WebKit WPE-based web browser source element
+
+- Video4Linux: HEVC encoding and decoding, JPEG encoding, and improved
+ dmabuf import/export
+
+- Hardware-accelerated Nvidia video decoder gained support for VP8/VP9
+ decoding, whilst the encoder gained support for H.265/HEVC encoding.
+
+- Many improvements to the Intel Media SDK based hardware-accelerated
+ video decoder and encoder plugin (msdk): dmabuf import/export for
+ zero-copy integration with other components; VP9 decoding; 10-bit
+ HEVC encoding; video post-processing (vpp) support including
+ deinterlacing; and the video decoder now handles dynamic resolution
+ changes.
+
+- The ASS/SSA subtitle overlay renderer can now handle multiple
+ subtitles that overlap in time and will show them on screen
+ simultaneously
+
+- The Meson build is now feature-complete (*) and it is now the
+ recommended build system on all platforms. The Autotools build is
+ scheduled to be removed in the next cycle.
+
+- The GStreamer Rust bindings and Rust plugins module are now
+ officially part of upstream GStreamer.
+
+- The GStreamer Editing Services gained a gesdemux element that allows
+ directly playing back serialized edit list with playbin or
+ (uri)decodebin
+
+- Many performance improvements
+
+
+Major new features and changes
+
+Noteworthy new API
+
+- GstAggregator has a new "min-upstream-latency" property that forces
+ a minimum aggregate latency for the input branches of an aggregator.
+ This is useful for dynamic pipelines where branches with a higher
+ latency might be added later after the pipeline is already up and
+ running and where a change in the latency would be disruptive. This
+ only applies to the case where at least one of the input branches is
+ live though, it won’t force the aggregator into live mode in the
+ absence of any live inputs.
+
+- GstBaseSink gained a "processing-deadline" property and
+ setter/getter API to configure a processing deadline for live
+ pipelines. The processing deadline is the acceptable amount of time
+ to process the media in a live pipeline before it reaches the sink.
+ This is on top of the systemic latency that is normally reported by
+ the latency query. This defaults to 20ms and should make pipelines
+ such as v4l2src ! xvimagesink not claim that all frames are late in
+ the QoS events. Ideally, this should replace the "max-lateness"
+ property for most applications.
+
+- RTCP Extended Reports (XR) parsing according to RFC 3611:
+ Loss/Duplicate RLE, Packet Receipt Times, Receiver Reference Time,
+ Delay since the last Receiver (DLRR), Statistics Summary, and VoIP
+ Metrics reports. This only provides the ability to parse such
+ packets, generation of XR packets is not supported yet and XR
+ packets are not automatically parsed by rtpbin / rtpsession but must
+ be actively handled by the application.
+
+- a new mode for interlaced video was added where each buffer carries
+ a single field of interlaced video, with buffer flags indicating
+ whether the field is the top field or bottom field. Top and bottom
+ fields are expected to alternate in this mode. Caps for this
+ interlace mode must also carry a format:Interlaced caps feature to
+ ensure backwards compatibility.
+
+- The video library has gained support for three new raw pixel
+ formats:
+
+ - Y410: packed 4:4:4 YUV, 10 bits per channel
+ - Y210: packed 4:2:2 YUV, 10 bits per channel
+ - NV12_10LE40: fully-packed 10-bit variant of NV12_10LE32,
+ i.e. without the padding bits
+
+- GstRTPSourceMeta is a new meta that can be used to transport
+ information about the origin of depayloaded or decoded RTP buffers,
+ e.g. when mixing audio from multiple sources into a single stream. A
+ new "source-info" property on the RTP depayloader base class
+ determines whether depayloaders should put this meta on outgoing
+ buffers. Similarly, the same property on RTP payloaders determines
+ whether they should use the information from this meta to construct
+ the CSRCs list on outgoing RTP buffers.
+
+- gst_sdp_message_from_text() is a convenience constructor to parse
+ SDPs from a string which is particularly useful for language
+ bindings.
+
+Support for Planar (Non-Interleaved) Raw Audio
+
+Raw audio samples are usually passed around in interleaved form in
+GStreamer, which means that if there are multiple audio channels the
+samples for each channel are interleaved in memory, e.g.
+|LEFT|RIGHT|LEFT|RIGHT|LEFT|RIGHT| for stereo audio. A non-interleaved
+or planar arrangement in memory would look like
+|LEFT|LEFT|LEFT|RIGHT|RIGHT|RIGHT| instead, possibly with
+|LEFT|LEFT|LEFT| and |RIGHT|RIGHT|RIGHT| residing in separate memory
+chunks or separated by some padding.
+
+GStreamer has always had signalling for non-interleaved audio since
+version 1.0, but it was never actually properly implemented in any
+elements. audioconvert would advertise support for it, but wasn’t
+actually able to handle it correctly.
+
+With this release we now have full support for non-interleaved audio as
+well, which means more efficient integration with external APIs that
+handle audio this way, but also more efficient processing of certain
+operations like interleaving multiple 1-channel streams into a
+multi-channel stream which can be done without memory copies now.
+
+New API to support this has been added to the GStreamer Audio support
+library: There is now a new GstAudioMeta which describes how data is
+laid out inside the buffer, and buffers with non-interleaved audio must
+always carry this meta. To access the non-interleaved audio samples you
+must map such buffers with gst_audio_buffer_map() which works much like
+gst_buffer_map() or gst_video_frame_map() in that it will populate a
+little GstAudioBuffer helper structure passed to it with the number of
+samples, the number of planes and pointers to the start of each plane in
+memory. This function can also be used to map interleaved audio buffers
+in which case there will be only one plane of interleaved samples.
+
+Of course support for this has also been implemented in the various
+audio helper and conversion APIs, base classes, and in elements such as
+audioconvert, audioresample, audiotestsrc, audiorate.
+
+Support for Closed Captions and Other Ancillary Data in Video
+
+The video support library has gained support for detecting and
+extracting Ancillary Data from videos as per the SMPTE S291M
+specification, including:
+
+- 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
+
+- line21encoder, line21decoder: inject/extract line21 closed captions
+ to/from SD video streams
+
+- cc708overlay: decodes CEA 608/708 captions and overlays them on
+ video
+
+Additionally, the following elements have also gained Closed Caption
+support:
+
+- qtdemux and qtmux support CEA 608/708 Closed Caption tracks
+
+- mpegvideoparse, h264parse extracts Closed Captions from MPEG-2/H.264
+ video streams
+
+- avviddec, avvidenc, x264enc got support for extracting/injecting
+ Closed Captions
+
+- decklinkvideosink can output closed captions and decklinkvideosrc
+ can extract closed captions
+
+- playbin and playbin3 learned how to autoplug CEA 608/708 CC overlay
+ elements
+
+- the externally maintained ajavideosrc element for AJA capture cards
+ has support for extracting closed captions
+
+The rsclosedcaption plugin in the Rust plugins collection includes a
+MacCaption (MCC) file parser and encoder.
+
+New Elements
+
+- overlaycomposition: New element that allows applications to draw
+ GstVideoOverlayCompositions on a stream. The element will emit the
+ "draw" signal for each video buffer, and the application then
+ generates an overlay for that frame (or not). This is much more
+ performant than e.g. cairooverlay for many use cases, e.g. because
+ pixel format conversions can be avoided or the blitting of the
+ overlay can be delegated to downstream elements (such as
+ gloverlaycompositor). It’s particularly useful for cases where only
+ a small section of the video frame should be drawn on.
+
+- gloverlaycompositor: New OpenGL-based compositor element that
+ flattens any overlays from GstVideoOverlayCompositionMetas into the
+ video stream. This element is also always part of glimagesink.
+
+- glalpha: New element that adds an alpha channel to a video stream.
+ The values of the alpha channel can either be set to a constant or
+ can be dynamically calculated via chroma keying. It is similar to
+ the existing alpha element but based on OpenGL. Calculations are
+ done in floating point so results may not be identical to the output
+ of the existing alpha element.
+
+- rtpfunnel funnels together RTP streams into a single session. Use
+ cases include multiplexing and bundle. webrtcbin uses it to
+ implement BUNDLE support.
+
+- testsrcbin is a source element that provides an audio and/or video
+ stream and also announces them using the recently-introduced
+ GstStream API. This is useful for testing elements such as playbin3
+ or uridecodebin3 etc.
+
+- New closed caption elements: cccombiner, ccextractor, ccconverter,
+ line21encoder, line21decoder and cc708overlay (see above)
+
+- wpesrc: new source element acting as a Web Browser based on WebKit
+ WPE
+
+- Two new OpenCV-based elements: cameracalibrate and cameraundistort
+ that can communicate to figure out distortion correction parameters
+ for a camera and correct for the distortion.
+
+- New sctp plugin based on usrsctp with sctpenc and sctpdec elements.
+ These elements are used inside webrtcbin for implementing data
+ channels.
+
+New element features and additions
+
+- playbin3, playbin and playsink have gained a new "text-offset"
+ property to adjust the positioning of the selected subtitle stream
+ vis-a-vis the audio and video streams. This uses subtitleoverlay’s
+ new "subtitle-ts-offset" property. GstPlayer has gained matching API
+ for this, namely gst_player_get_text_video_offset().
+
+- playbin3 buffering improvements: in network playback scenarios there
+ may be multiple inputs to decodebin3, and buffering will be done
+ before decodebin3 using queue2 or downloadbuffer elements inside
+ urisourcebin. Since this is before any parsers or demuxers there may
+ not be any bitrate information available for the various streams, so
+ it was difficult to configure the buffering there smartly within
+ global constraints. This was improved now: The queue2 elements
+ inside urisourcebin will now use the new bitrate query to figure out
+ a bitrate estimate for the stream if no bitrate was provided by
+ upstream, and urisourcebin will use the bitrates of the individual
+ queues to distribute the globally-set "buffer-size" budget in bytes
+ to the various queues. urisourcebin also gained "low-watermark" and
+ "high-watermark" properties which will be proxied to the internal
+ queues, as well as a read-only "statistics" property which allows
+ querying of the minimum/maximum/average byte and time levels of the
+ queues inside the urisourcebin in question.
+
+- splitmuxsink has gained a couple of new features:
+
+ - new "async-finalize" mode: This mode is useful for muxers or
+ outputs that can take a long time to finalize a file. Instead of
+ blocking the whole upstream pipeline while the muxer is doing
+ its stuff, we can unlink it and spawn a new muxer + sink
+ combination to continue running normally. This requires us to
+ receive the muxer and sink (if needed) as factories via the new
+ "muxer-factory" and "sink-factory" properties, optionally
+ accompanied by their respective properties structures (set via
+ the new "muxer-properties" and "sink-properties" properties).
+ There are also new "muxer-added" and "sink-added" signals in
+ case custom code has to be called for them to configure them.
+
+ - "split-at-running-time" action signal: When called by the user,
+ this action signal ends the current file (and starts a new one)
+ as soon as the given running time is reached. If called multiple
+ times, running times are queued up and processed in the order
+ they were given.
+
+ - "split-after" action signal to finish outputting the current GOP
+ to the current file and then start a new file as soon as the GOP
+ is finished and a new GOP is opened (unlike the existing
+ "split-now" which immediately finishes the current file and
+ writes the current GOP into the next newly-started file).
+
+ - "reset-muxer" property: when unset, the muxer is reset using
+ flush events instead of setting its state to NULL and back. This
+ means the muxer can keep state across resets, e.g. mpegtsmux
+ will keep the continuity counter continuous across segments as
+ required by hlssink2.
+
+- qtdemux gained PIFF track encryption box support in addition to the
+ already-existing PIFF sample encryption support, and also allows
+ applications to select which encryption system to use via a
+ "drm-preferred-decryption-system-id" context in case there are
+ multiple options.
+
+- qtmux: the "start-gap-threshold" property determines now whether an
+ edit list will be created to account for small gaps or offsets at
+ the beginning of a stream in case the start timestamps of tracks
+ don’t line up perfectly. Previously the threshold was hard-coded to
+ 1% of the (video) frame duration, now it is 0 by default (so edit
+ list will be created even for small differences), but fully
+ configurable.
+
+- rtpjitterbuffer has improved end-of-stream handling
+
+- rtpmp4vpay will be prefered over rtpmp4gpay for MPEG-4 video in
+ autoplugging scenarios now
+
+- rtspsrc now allows applications to send RTSP SET_PARAMETER and
+ GET_PARAMETER requests using action signals.
+
+- rtspsrc has a small (100ms) configurable teardown delay by default
+ to try and make sure an RTSP TEARDOWN request gets sent out when the
+ source element shuts down. This will block the downward PAUSED to
+ READY state change for a short time, but can be disabled where it’s
+ a problem. Some servers only allow a limited number of concurrent
+ clients, so if no proper TEARDOWN is sent new clients may have
+ problems connecting to the server for a while.
+
+- souphttpsrc behaves better with low bitrate streams now. Before it
+ would increase the read block size too quickly which could lead to
+ it not reading any data from the socket for a very long time with
+ low bitrate streams that are output live downstream. This could lead
+ to servers kicking off the client.
+
+- filesink: do internal buffering to avoid performance regression with
+ small writes since we bypass libc buffering by using writev()
+ instead of fwrite()
+
+- identity: add "eos-after" property and fix "error-after" property
+ when the element is reused
+
+- input-selector: lets context queries pass through, so that
+ e.g. upstream OpenGL elements can use contexts and displays
+ advertised by downstream elements
+
+- queue2: avoid ping-pong between 0% and 100% buffering messages if
+ upstream is pushing buffers larger than one of its limits, plus
+ performance optimisations
+
+- opusdec: new "phase-inversion" property to control phase inversion.
+ When enabled, this will slightly increase stereo quality, but
+ produces a stream that when downmixed to mono will suffer audio
+ distortions.
+
+- The x265enc HEVC encoder also exposes a "key-int-max" property to
+ configure the maximum allowed GOP size now.
+
+- decklinkvideosink has seen stability improvements for long-running
+ pipelines (potential crash due to overflow of leaked clock refcount)
+ and clock-slaving improvements when performing flushing seeks
+ (causing stalls in the output timeline), pausing and/or buffering.
+
+- srtpdec, srtpenc: add support for MKIs which allow multiple keys to
+ be used with a single SRTP stream
+
+- srtpdec, srtpenc: add support for AES-GCM and also add support for
+ it in gst-rtsp-server and rtspsrc.
+
+- The srt Secure Reliable Transport plugin has integrated server and
+ client elements srt{client,server}{src,sink} into one (srtsrc and
+ srtsink), since SRT connection mode can be changed by uri
+ parameters.
+
+- h264parse and h265parse will handle SEI recovery point messages and
+ mark recovery points as keyframes as well (in addition to IDR
+ frames)
+
+- webrtcbin: "add-turn-server" action signal to pass multiple ICE
+ relays (TURN servers).
+
+- The removesilence element has received various new features and
+ properties, such as a "threshold" property, detecting silence only
+ after minimum silence time/buffers, a "silent" property to control
+ bus message notifications as well as a "squash" property.
+
+- AOMedia AV1 decoder gained support for 10/12bit decoding whilst the
+ AV1 encoder supports more image formats and subsamplings now and
+ acquired support for rate control and profile related configuration.
+
+- The Fraunhofer fdkaac plugin can now be built against the 2.0.0
+ version API and has improved multichannel support
+
+- kmssink now supports unpadded 24-bit RGB and can configure mode
+ setting from video info, which enables display of multi-planar
+ formats such as I420 or NV12 with modesetting. It has also gained a
+ number of new properties: The "restore-crtc" property does what it
+ says on the tin and is enabled by default. "plane-properties" and
+ "connector-properties" can be used to pass custom properties to the
+ DRM.
+
+- waylandsink has a "fullscreen" property now and supports the
+ XDG-Shell protocol.
+
+- decklinkvideosink, decklinkvideosrc support selecting between
+ half/full duplex
+
+- The vulkan plugin gained support for macOS and iOS via MoltenVK in
+ addition to the existing support for X11 and Wayland
+
+- imagefreeze has a new num-buffers property to limit the number of
+ buffers that are produced and to send an EOS event afterwards
+
+- webrtcbin has a new, introspectable get-transceiver signal in
+ addition to the old get-transceivers signal that couldn’t be used
+ from bindings
+
+- Support for per-element latency information was added to the latency
+ tracer
+
+Plugin and library moves
+
+- The stereo element was moved from -bad into the existing audiofx
+ plugin in -good. If you get duplicate type registration warnings
+ when upgrading, check that you don’t have a stale stereoplugin lying
+ about somewhere.
+
+GstVideoAggregator, compositor, and OpenGL mixer elements moved from -bad to -base
+
+GstVideoAggregator is a new base class for raw video mixers and muxers
+and is based on GstAggregator. It provides defined-latency mixing of raw
+video inputs and ensures that the pipeline won’t stall even if one of
+the input streams stops producing data.
+
+As part of the move to stabilise the API there were some last-minute API
+changes and clean-ups, but those should mostly affect internal elements.
+Most notably, the "ignore-eos" pad property was renamed to
+"repeat-after-eos" and the conversion code was moved to a
+GstVideoAggregatorConvertPad subclass to avoid code duplication, make
+things less awkward for subclasses like the OpenGL-based video mixer,
+and make the API more consistent with the audio aggregator API.
+
+It is used by the compositor element, which is a replacement for
+‘videomixer’ which did not handle live inputs very well. compositor
+should behave much better in that respect and generally behave as one
+would expected in most scenarios.
+
+The compositor element has gained support for per-pad blending mode
+operators (SOURCE, OVER, ADD) which determines what operator to use for
+blending this pad over the previous ones. This can be used to implement
+crossfading and the available operators can be extended in the future as
+needed.
+
+A number of OpenGL-based video mixer elements (glvideomixer, glmixerbin,
+glvideomixerelement, glstereomix, glmosaic) which are built on top of
+GstVideoAggregator have also been moved from -bad to -base now. These
+elements have been merged into the existing OpenGL plugin, so if you get
+duplicate type registration warnings when upgrading, check that you
+don’t have a stale openglmixers plugin lying about somewhere.
+
+Plugin removals
+
+The following plugins have been removed from gst-plugins-bad:
+
+- The experimental daala plugin has been removed, since it’s not so
+ useful now that all effort is focused on AV1 instead, and it had to
+ be enabled explicitly with --enable-experimental anyway.
+
+- The spc plugin has been removed. It has been replaced by the gme
+ plugin.
+
+- The acmmp3dec and acmenc plugins for Windows have been removed. ACM
+ is an ancient legacy API and there was no point in keeping the
+ plugins around for a licensed MP3 decoder now that the MP3 patents
+ have expired and we have a decoder in -good. We also didn’t ship
+ these in our cerbero-built Windows packages, so it’s unlikely that
+ they’ll be missed.
+
+
+Miscellaneous API additions
+
+- GstBitwriter: new generic bit writer API to complement the existing
+ bit reader
+
+- gst_buffer_new_wrapped_bytes() creates a wrap buffer from a GBytes
+
+- gst_caps_set_features_simple() sets a caps feature on all the
+ structures of a GstCaps
+
+- New GST_QUERY_BITRATE query: This allows determining from downstream
+ what the expected bitrate of a stream may be which is useful in
+ queue2 for setting time based limits when upstream does not provide
+ timing information. tsdemux, qtdemux and matroskademux have basic
+ support for this query on their sink pads.
+
+- elements: there is a new “Hardware” class specifier. Elements
+ interacting with hardware devices should specify this classifier in
+ their element factory class metadata. This is useful to advertise as
+ one might need to put such elements into READY state to test if the
+ hardware is present in the system for example.
+
+- protection: Add a new definition for unspecified system protection,
+ GST_PROTECTION_UNSPECIFIED_SYSTEM_ID
+
+- take functions for various mini objects that didn’t have them yet:
+ gst_query_take(), gst_message_take(), gst_tag_list_take(),
+ gst_buffer_list_take(). Unlike the various _replace() functions
+ _take() does not increase the reference count but takes ownership of
+ the mini object passed.
+
+- clear functions for various mini object types and GstObject which
+ unrefs the object or mini object (if non-NULL) and sets the variable
+ pointed to to NULL: gst_clear_structure(), gst_clear_tag_list(),
+ gst_clear_query(), gst_clear_message(), gst_clear_event(),
+ gst_clear_caps(), gst_clear_buffer_list(), gst_clear_buffer(),
+ gst_clear_mini_object(), gst_clear_object()
+
+- miniobject: new API gst_mini_object_add_parent() and
+ gst_mini_object_remove_parent() to set parent pointers on mini
+ objects to ensure correct writability: Every container of
+ miniobjects now needs to store itself as parent in the child object,
+ and remove itself again later. A mini object is then only writable
+ if there is at most one parent, that parent is writable itself, and
+ the reference count of the mini object is 1. GstBuffer (for
+ memories), GstBufferList (for buffers), GstSample (for caps, buffer,
+ bufferlist), and GstVideoOverlayComposition were updated
+ accordingly. Without this it was possible to have e.g. a buffer list
+ with a refcount of 2 used in two places at once that both modify the
+ same buffer with refcount 1 at the same time wrongly thinking it is
+ writable even though it’s really not.
+
+- poll: add API to watch for POLLPRI and stop treating POLLPRI as a
+ read. This is useful to wait for video4linux events which are
+ signalled via POLLPRI.
+
+- sample: new API to update the contents of a GstSample and make it
+ writable: gst_sample_set_buffer(), gst_sample_set_caps(),
+ gst_sample_set_segment(), gst_sample_set_info(), plus
+ gst_sample_is_writable() and gst_sample_make_writable(). This makes
+ it possible to reuse a sample object and avoid unnecessary memory
+ allocations, for example in appsink.
+
+- ClockIDs now keep a weak reference to underlying clock to avoid
+ crashes in basesink in corner cases where a clock goes away while
+ the ClockID is still in use, plus some new API
+ (gst_clock_id_get_clock(), gst_clock_id_uses_clock()) to check the
+ clock a ClockID is linked to.
+
+- The GstCheck unit test library gained a
+ fail_unless_equals_clocktime() convenience macro as well as some new
+ GstHarness API for for proposing meta APIs from the allocation
+ query: gst_harness_add_propose_allocation_meta(). ASSERT_CRITICAL()
+ checks in unit tests are now skipped if GStreamer was compiled with
+ GST_DISABLE_GLIB_CHECKS.
+
+- gst_audio_buffer_truncate() convenience function to truncate a raw
+ audio buffer
+
+- GstDiscoverer has support for caching the results of discovery in
+ the default cache directory. This can be enabled with the use-cache
+ property and is disabled by default.
+
+- GstMeta that are attached to GstBuffers are now always stored in the
+ order in which they were added.
+
+- Additional support for signalling ONVIF specific features were
+ added: the SEEK event can store a trickmode-interval now and support
+ for the Rate-Control and Frames RTSP headers was added to the RTSP
+ library.
+
+
+Miscellaneous performance and memory optimisations
+
+As always there have been many performance and memory usage improvements
+across all components and modules. Some of them (such as dmabuf
+import/export) have already been mentioned elsewhere so won’t be
+repeated here.
+
+The following list is only a small snapshot of some of the more
+interesting optimisations that haven’t been mentioned in other contexts
+yet:
+
+- The GstVideoEncoder and GstVideoDecoder base classes now release the
+ STREAM_LOCK when pushing out buffers, which means (multi-threaded)
+ encoders and decoders can now receive and continue to process input
+ buffers whilst waiting for downstream elements in the pipeline to
+ process the buffer that was pushed out. This increases throughput
+ and reduces processing latency, also and especially for
+ hardware-accelerated encoder/decoder elements.
+
+- GstQueueArray has seen a few API additions
+ (gst_queue_array_peek_nth(), gst_queue_array_set_clear_func(),
+ gst_queue_array_clear()) so that it can be used in other places like
+ GstAdapter instead of a GList, which reduces allocations and
+ improves performance.
+
+- appsink now reuses the sample object in pull_sample() if possible
+
+- rtpsession only starts the RTCP thread when it’s actually needed now
+
+- udpsrc uses a buffer pool now and the GstUdpSrc object structure was
+ optimised for better cache performance
+
+GstPlayer
+
+- API was added to fine-tune the synchronisation offset between
+ subtitles and video
+
+
+Miscellaneous changes
+
+- As a result of moving to newer FFmpeg APIs, encoder and decoder
+ elements exposed by the GStreamer FFmpeg wrapper plugin (gst-libav)
+ may have seen possibly incompatible changes to property names and/or
+ types, and not all properties exposed might be functional. We are
+ still reviewing the new properties and aim to minimise breaking
+ changes at least for the most commonly-used properties, so please
+ report any issues you run into!
+
+OpenGL integration
+
+- The OpenGL mixer elements have been moved from -bad to
+ gst-plugins-base (see above)
+
+- The Mesa GBM backend now supports headless mode
+
+- gloverlaycompositor: New OpenGL-based compositor element that
+ flattens any overlays from GstVideoOverlayCompositionMetas into the
+ video stream.
+
+- glalpha: New element that adds an alpha channel to a video stream.
+ The values of the alpha channel can either be set to a constant or
+ can be dynamically calculated via chroma keying. It is similar to
+ the existing alpha element but based on OpenGL. Calculations are
+ done in floating point so results may not be identical to the output
+ of the existing alpha element.
+
+- glupload: Implement direct dmabuf uploader, the idea being that some
+ GPUs (like the Vivante series) can actually perform the YUV->RGB
+ conversion internally, so no custom conversion shaders are needed.
+ To make use of this feature, we need an additional uploader that can
+ import DMABUF FDs and also directly pass the pixel format, relying
+ on the GPU to do the conversion.
+
+- The OpenGL library no longer restores the OpenGL viewport. This is a
+ performance optimization to not require performing multiple
+ expensive glGet*() function calls per frame. This affects any
+ application or plugin use of the following functions and objects:
+ - glcolorconvert library object (not the element)
+ - glviewconvert library object (not the element)
+ - gst_gl_framebuffer_draw_to_texture()
+ - custom GstGLWindow implementations
+
+
+Tracing framework and debugging improvements
+
+- There is now a GDB PRETTY PRINTER FOR VARIOUS GSTREAMER TYPES: For
+ GstObject pointers the type and name is added, e.g.
+ 0x5555557e4110 [GstDecodeBin|decodebin0]. For GstMiniObject pointers
+ the object type is added, e.g. 0x7fffe001fc50 [GstBuffer]. For
+ GstClockTime and GstClockTimeDiff the time is also printed in human
+ readable form, e.g. 150116219955 [+0:02:30.116219955].
+
+- GDB EXTENSION WITH TWO CUSTOM GDB COMMANDS gst-dot AND gst-print:
+
+ - gst-dot creates dot files that a very close to what
+ GST_DEBUG_BIN_TO_DOT_FILE() produces, but object properties and
+ buffer contents such as codec-data in caps are not available.
+
+ - gst-print produces high-level information about a GStreamer
+ object. This is currently limited to pads for GstElements and
+ events for the pads. The output may look like this:
+
+- gst_structure_to_string() now serialises the actual value of
+ pointers when serialising GstStructures instead of claiming they’re
+ NULL. This makes debug logging in various places less confusing,
+ because it’s clear now that structure fields actually hold valid
+ objects. Such object pointer values will never be deserialised
+ however.
+
+
+Tools
+
+- gst-inspect-1.0 has coloured output now and will automatically use a
+ pager if the output does not fit on a page. This only works in a
+ UNIX environment and if the output is not piped, and on Windows 10
+ build 16257 or newer. If you don’t like the colours you can disable
+ them by setting the GST_INSPECT_NO_COLORS=1 environment variable or
+ passing the --no-color command line option.
+
+
+GStreamer RTSP server
+
+- Improved backlog handling when using TCP interleaved for data
+ transport. Before there was a fixed maximum size for backlog
+ messages, which was prone to deadlocks and made it difficult to
+ control memory usage with the watch backlog. The RTSP server now
+ limits queued TCP data messages to one per stream, moving queuing of
+ the data into the pipeline and leaving the RTSP connection
+ responsive to RTSP messages in both directions, preventing all those
+ problems.
+
+- Initial ULP Forward Error Correction support in rtspclientsink and
+ for RECORD mode in the server.
+
+- API to explicitly enable retransmission requests (RTX)
+
+- Lots of multicast-related fixes
+
+- rtsp-auth: Add support for parsing .htdigest files
+
+
+GStreamer VAAPI
+
+- Support Wayland’s display for context sharing, so the application
+ can pass its own wl_display in order to be used for the VAAPI
+ display creation.
+
+- A lot of work to support new Intel hardware using media-driver as VA
+ backend.
+
+- For non-x86 devices, VAAPI display can instantiate, through DRM,
+ with no PCI bus. This enables the usage of libva-v4l2-request
+ driver.
+
+- Added support for XDG-shell protocol as wl_shell replacement which
+ is currently deprecated. This change add as dependency
+ wayland-protocol.
+
+- GstVaapiFilter, GstVaapiWindow, and GstVaapiDecoder classes now
+ inherit from GstObject, gaining all the GStreamer’s instrumentation
+ support.
+
+- The metadata now specifies the plugin as Hardware class.
+
+- H264 decoder is more stable with problematic streams.
+
+- In H265 decoder added support for profiles main-422-10 (P010_10LE),
+ main-444 (AYUV) and main-444-10 (Y410)
+
+- JPEG decoder handles dynamic resolution changes.
+
+- More specification adherence in H264 and H265 encoders.
+
+
+GStreamer OMX
+
+- Add support of NV16 format to video encoders input.
+
+- Video decoders now handle the ALLOCATION query to tell upstream
+ about the number of buffers they require. Video encoders will also
+ use this query to adjust their number of allocated buffers
+ preventing starvation when using dynamic buffer mode.
+
+- The OMX_PERFORMANCE debug category has been renamed to OMX_API_TRACE
+ and can now be used to track a widder variety of interactions
+ between OMX and GStreamer.
+
+- Video encoders will now detect frame rate only changes and will
+ inform OMX about it rather than doing a full format reset.
+
+- Various Zynq UltraScale+ specific improvements:
+ - Video encoders are now able to import dmabuf from upstream.
+ - Support for HEVC range extension profiles and more AVC profiles.
+ - We can now request video encoders to generate an IDR using the
+ force key unit event.
+
+
+GStreamer Editing Services and NLE
+
+- Added a gesdemux element, it is an auto pluggable element that
+ allows decoding edit list like files supported by GES
+
+- Added gessrc which wraps a GESTimeline as a standard source element
+ (implementing the ges protocol handler)
+
+- Added basic support for videorate::rate property potentially
+ allowing changing playback speed
+
+- Layer priority is now fully automatic and they should be moved with
+ the new ges_timeline_move_layer method, ges_layer_set_priority is
+ now deprecated.
+
+- Added a ges_timeline_element_get_layer_priority so we can simply get
+ all information about GESTimelineElement position in the timeline
+
+- GESVideoSource now auto orientates the images if it is defined in a
+ meta (overridable).
+
+- Added some PyGObject overrides to make the API more pythonic
+
+- The threading model has been made more explicit with safe guard to
+ make sure not thread safe APIs are not used from the wrong threads.
+ It is also now possible to properly handle in what thread the API
+ should be used.
+
+- Optimized GESClip and GESTrackElement creation
+
+- Added a way to compile out the old, unused and deprecated
+ GESPitiviFormatter
+
+- Re implemented the timeline editing API making it faster and making
+ the code much more maintainable
+
+- Simplified usage of nlecomposition outside GES by removing quirks in
+ it API usage and removing the need to treat it specially from an
+ application perspective.
+
+- ges-launch-1.0:
+
+ - Added support to add titles to the timeline
+ - Enhance the help auto generating it from the code
+
+- Deprecate ges_timeline_load_from_uri as loading the timeline should
+ be done through a project now
+
+- MANY leaks have been plugged and the unit testsuite is now “leak
+ free”
+
+
+GStreamer validate
+
+- Added an action type to verify the checksum of the sink last-sample
+
+- Added an include keyword to validate scenarios
+
+- Added the notion of variable in scenarios, with the set-vars keyword
+
+- Started adding support for “performance” like tests by allowing to
+ define the number of dropped buffers or the minimum buffer frequency
+ on a specific pad
+
+- Added a validateflow plugin which allows defining the data flow to
+ be seen on a particular pad and verifying that following runs match
+ the expectations
+
+- Added support for appsrc based test definition so we can instrument
+ the data pushed into the pipeline from scenarios
+
+- Added a mockdecryptor allowing adding tests with on encrypted files,
+ the element will potentially be instrumented with a validate
+ scenario
+
+- gst-validate-launcher:
+
+ - Cleaned up output
+
+ - Changed the default for “muting” tests as user doesn’t expect
+ hundreds of windows to show up when running the testsuite
+
+ - Fixed the outputted xunit files to be compatible with GitLab
+
+ - Added support to run tests on media files in push mode (using
+ pushfile://)
+
+ - Added support for running inside gst-build
+
+ - Added support for running ssim tests on rendered files
+
+ - Added a way to simply define tests on pipelines through a simple
+ .json file
+
+ - Added a python app to easily run python testsuite reusing all
+ the launcher features
+
+ - Added flatpak knowledge so we can print backtrace even when
+ running from within flatpak
+
+ - Added a way to automatically generated “known issues”
+ suppressions lines
+
+ - Added a way to rerun tests to check if they are flaky and added
+ a way to tolerate tests known to be flaky
+
+ - Add a way to output html log files
+
+
+GStreamer Python Bindings
+
+- add binding for gst_pad_set_caps()
+
+- pygobject dependency requirement was bumped to >= 3.8
+
+- new audiotestsrc, audioplot, and mixer plugin examples, and a
+ dynamic pipeline example
+
+
+GStreamer C# Bindings
+
+- bindings for the GstWebRTC library
+
+
+GStreamer Rust Bindings
+
+The GStreamer Rust bindings are now officially part of the GStreamer
+project and are also maintained in the GStreamer GitLab.
+
+The releases will generally not be synchronized with the releases of
+other GStreamer parts due to dependencies on other projects.
+
+Also unlike the other GStreamer libraries, the bindings will not commit
+to full API stability but instead will follow the approach that is
+generally taken by Rust projects, e.g.:
+
+1) 0.12.X will be completely API compatible with all other 0.12.Y
+ versions.
+2) 0.12.X+1 will contain bugfixes and compatible new feature additions.
+3) 0.13.0 will _not_ be backwards compatible with 0.12.X but projects
+ will be able to stay at 0.12.X without any problems as long as they
+ don’t need newer features.
+
+The current stable release is 0.12.2 and the next release series will be
+0.13, probably around March 2019.
+
+At this point the bindings cover most of GStreamer core (except for most
+notably GstAllocator and GstMemory), and most parts of the app, audio,
+base, check, editing-services, gl, net. pbutils, player, rtsp,
+rtsp-server, sdp, video and webrtc libraries.
+
+Also included is support for creating subclasses of the following types
+and writing GStreamer plugins:
+
+- gst::Element
+- gst::Bin and gst::Pipeline
+- gst::URIHandler and gst::ChildProxy
+- gst::Pad, gst::GhostPad
+- gst_base::Aggregator and gst_base::AggregatorPad
+- gst_base::BaseSrc and gst_base::BaseSink
+- gst_base::BaseTransform
+
+Changes to 0.12.X since 0.12.0
+
+Fixed
+
+- PTP clock constructor actually creates a PTP instead of NTP clock
+
+Added
+
+- Bindings for GStreamer Editing Services
+- Bindings for GStreamer Check testing library
+- Bindings for the encoding profile API (encodebin)
+
+- VideoFrame, VideoInfo, AudioInfo, StructureRef implements Send and
+ Sync now
+- VideoFrame has a function to get the raw FFI pointer
+- From impls from the Error/Success enums to the combined enums like
+ FlowReturn
+- Bin-to-dot file functions were added to the Bin trait
+- gst_base::Adapter implements SendUnique now
+- More complete bindings for the gst_video::VideoOverlay interface,
+ especially
+ gst_video::is_video_overlay_prepare_window_handle_message()
+
+Changed
+
+- All references were updated from GitHub to freedesktop.org GitLab
+- Fix various links in the README.md
+- Link to the correct location for the documentation
+- Remove GitLab badge as that only works with gitlab.com currently
+
+Changes in git master for 0.13
+
+Fixed
+
+- gst::tag::Album is the album tag now instead of artist sortname
+
+Added
+
+- Subclassing infrastructure was moved directly into the bindings,
+ making the gst-plugin crate deprecated. This involves many API
+ changes but generally cleans up code and makes it more flexible.
+ Take a look at the gst-plugins-rs crate for various examples.
+
+- Bindings for CapsFeatures and Meta
+- Bindings for
+ ParentBufferMeta,VideoMetaandVideoOverlayCompositionMeta`
+- Bindings for VideoOverlayComposition and VideoOverlayRectangle
+- Bindings for VideoTimeCode
+
+- UniqueFlowCombiner and UniqueAdapter wrappers that make use of the
+ Rust compile-time mutability checks and expose more API in a safe
+ way, and as a side-effect implement Sync and Send now
+
+- More complete bindings for Allocation Query
+- pbutils functions for codec descriptions
+- TagList::iter() for iterating over all tags while getting a single
+ value per tag. The old ::iter_tag_list() function was renamed to
+ ::iter_generic() and still provides access to each value for a tag
+- Bus::iter() and Bus::iter_timed() iterators around the corresponding
+ ::pop\*() functions
+
+- serde serialization of Value can also handle Buffer now
+
+- Extensive comments to all examples with explanations
+- Transmuxing example showing how to use typefind, multiqueue and
+ dynamic pads
+- basic-tutorial-12 was ported and added
+
+Changed
+
+- Rust 1.31 is the minimum supported Rust version now
+- Update to latest gir code generator and glib bindings
+
+- Functions returning e.g. gst::FlowReturn or other “combined” enums
+ were changed to return split enums like
+ Result<gst::FlowSuccess, gst::FlowError> to allow usage of the
+ standard Rust error handling.
+
+- MiniObject subclasses are now newtype wrappers around the underlying
+ GstRc<FooRef> wrapper. This does not change the API in any breaking
+ way for the current usages, but allows MiniObjects to also be
+ implemented in other crates and makes sure rustdoc places the
+ documentation in the right places.
+
+- BinExt extension trait was renamed to GstBinExt to prevent conflicts
+ with gtk::Bin if both are imported
+
+- Buffer::from_slice() can’t possible return None
+
+- Various clippy warnings
+
+
+GStreamer Rust Plugins
+
+Like the GStreamer Rust bindings, the Rust plugins are now officially
+part of the GStreamer project and are also maintained in the GStreamer
+GitLab.
+
+In the 0.3.x versions this contained infrastructure for writing
+GStreamer plugins in Rust, and a set of plugins.
+
+In git master that infrastructure was moved to the GLib and GStreamer
+bindings directly, together with many other improvements that were made
+possible by this, so the gst-plugins-rs repository only contains
+GStreamer elements now.
+
+Elements included are:
+
+- Tutorials plugin: identity, rgb2gray and sinesrc with extensive
+ comments
+
+- rsaudioecho, a port of the audiofx element
+
+- rsfilesrc, rsfilesink
+
+- rsflvdemux, a FLV demuxer. Not feature-equivalent with flvdemux yet
+
+- threadshare plugin: ts-appsrc, ts-proxysrc/sink, ts-queue, ts-udpsrc
+ and ts-tcpclientsrc elements that use a fixed number of threads and
+ share them between instances. For more background about these
+ elements see Sebastian’s talk “When adding more threads adds more
+ problems - Thread-sharing between elements in GStreamer” at the
+ GStreamer Conference 2017.
+
+- rshttpsrc, a HTTP source around the hyper/reqwest Rust libraries.
+ Not feature-equivalent with souphttpsrc yet.
+
+- togglerecord, an element that allows to start/stop recording at any
+ time and keeps all audio/video streams in sync.
+
+- mccparse and mccenc, parsers and encoders for the MCC closed caption
+ file format.
+
+Changes to 0.3.X since 0.3.0
+
+- All references were updated from GitHub to freedesktop.org GitLab
+- Fix various links in the README.md
+- Link to the correct location for the documentation
+
+Changes in git master for 0.4
+
+- togglerecord: Switch to parking_lot crate for mutexes/condition
+ variables for lower overhead
+- Merge threadshare plugin here
+- New closedcaption plugin with mccparse and mccenc elements
+- New identity element for the tutorials plugin
+
+- Register plugins statically in tests instead of relying on the
+ plugin loader to find the shared library in a specific place
+
+- Update to the latest API changes in the GLib and GStreamer bindings
+- Update to the latest versions of all crates
+
+
+Build and Dependencies
+
+- The MESON BUILD SYSTEM BUILD IS NOW FEATURE-COMPLETE (*) and it is
+ now the recommended build system on all platforms and also used by
+ Cerbero to build GStreamer on all platforms. The Autotools build is
+ scheduled to be removed in the next cycle. Developers who currently
+ use gst-uninstalled should move to gst-build. The build option
+ naming has been cleaned up and made consistent and there are now
+ feature options to enable/disable plugins and various other features
+ on a case-by-case basis. (*) with the exception of plugin docs which
+ will be handled differently in future
+
+- Symbol export in libraries is now controlled via explicit exports
+ using symbol visibility or export defines where supported, to ensure
+ consistency across all platforms. This also allows libraries to have
+ exports that vary based on detected platform features and configure
+ options as is the case with the GStreamer OpenGL integration library
+ for example. A few symbols that had been exported by accident in
+ earlier versions may no longer be exported. These symbols will not
+ have had declarations in any public header files then though and
+ would not have been usable.
+
+- The GStreamer FFmpeg wrapper plugin (gst-libav) now depends on
+ FFmpeg 4.x and uses the new FFmpeg 4.x API and stopped relying on
+ ancient API that was removed with the FFmpeg 4.x release. This means
+ that it is no longer possible to build this module against an older
+ system-provided FFmpeg 3.x version. Use the internal FFmpeg 4.x copy
+ instead if you build using autotools, or use gst-libav 1.14.x
+ instead which targets the FFmpeg 3.x API and _should_ work fine in
+ combination with a newer GStreamer. It’s difficult for us to support
+ both old and new FFmpeg APIs at the same time, apologies for any
+ inconvenience caused.
+
+- Hardware-accelerated Nvidia video encoder/decoder plugins nvdec and
+ nvenc can be built against CUDA Toolkit versions 9 and 10.0 now. The
+ dynlink interface has been dropped since it’s deprecated in 10.0.
+
+- The (optional) OpenCV requirement has been bumped to >= 3.0.0 and
+ the plugin can also be built against OpenCV 4.x now.
+
+- New sctp plugin based on usrsctp (for WebRTC data channels)
+
+Cerbero
+
+Cerbero is a meta build system used to build GStreamer plus dependencies
+on platforms where dependencies are not readily available, such as
+Windows, Android, iOS and macOS.
+
+Cerbero has seen a number of improvements:
+
+- Cerbero has been ported to Python 3 and requires Python 3.5 or newer
+ now
+
+- Source tarballs are now protected by checksums in the recipes to
+ guard against download errors and malicious takeover of projects or
+ websites. In addition, downloads are only allowed via secure
+ transports now and plain HTTP, FTP and git:// transports are not
+ allowed anymore.
+
+- There is now a new fetch-bootstrap command which downloads sources
+ required for bootstrapping, with an optional --build-tools-only
+ argument to match the bootstrap --build-tools-only command.
+
+- The bootstrap, build, package and bundle-source commands gained a
+ new --offline switch that ensures that only sources from the cache
+ are used and never downloaded via the network. This is useful in
+ combination with the fetch and fetch-bootstrap commands that acquire
+ sources ahead of time before any build steps are executed. This
+ allows more control over the sources used and when sources are
+ updated, and is particularly useful for build environments that
+ don’t have network access.
+
+- bootstrap --assume-yes will automatically say ‘yes’ to any
+ interactive prompts during the bootstrap stage, such as those from
+ apt-get or yum.
+
+- bootstrap --system-only will only bootstrap the system without build
+ tools.
+
+- Manifest support: The build manifest can be used in continuous
+ integration (CI) systems to fixate the Git revision of certain
+ projects so that all builds of a pipeline are on the same reference.
+ This is used in GStreamer’s gitlab CI for example. It can also be
+ used in order to re-produce a specific build. To set a manifest, you
+ can set manifest = 'my_manifest.xml' in your configuration file, or
+ use the --manifest command line option. The command line option will
+ take precendence over anything specific in the configuration file.
+
+- The new build-deps command can be used to build only the
+ dependencies of a recipe, without the recipe itself.
+
+- new --list-variants command to list available variants
+
+- variants can now be set on the command line via the -v option as a
+ comma-separated list. This overrides any variants set in any
+ configuration files.
+
+- new qt5, intelmsdk and nvidia variants for enabling Qt5 and hardware
+ codec support. See the Enabling Optional Features with Variants
+ section in the Cerbero documentation for more details how to enable
+ and use these variants.
+
+- 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
+
+Aaron Boxer, Aleix Conchillo Flaqué, Alessandro Decina, Alexandru Băluț,
+Alex Ashley, Alexey Chernov, Alicia Boya García, Amit Pandya, Andoni
+Morales Alastruey, Andreas Frisch, Andre McCurdy, Andy Green, Anthony
+Violo, Antoine Jacoutot, Antonio Ospite, Arun Raghavan, Aurelien Jarno,
+Aurélien Zanelli, ayaka, Bananahemic, Bastian Köcher, Branko Subasic,
+Brendan Shanks, Carlos Rafael Giani, Charlie Turner, Christoph Reiter,
+Corentin Noël, Daeseok Youn, Damian Vicino, Dan Kegel, Daniel Drake,
+Daniel Klamt, Danilo Spinella, Dardo D Kleiner, David Ing, David
+Svensson Fors, Devarsh Thakkar, Dimitrios Katsaros, Edward Hervey,
+Emilio Pozuelo Monfort, Enrique Ocaña González, Erlend Eriksen, Ezequiel
+Garcia, Fabien Dessenne, Fabrizio Gennari, Florent Thiéry, Francisco
+Velazquez, Freyr666, Garima Gaur, Gary Bisson, George Kiagiadakis, Georg
+Lippitsch, Georg Ottinger, Geunsik Lim, Göran Jönsson, Guillaume
+Desmottes, H1Gdev, Haihao Xiang, Haihua Hu, Harshad Khedkar, Havard
+Graff, He Junyan, Hoonhee Lee, Hosang Lee, Hyunjun Ko, Ilya Smelykh,
+Ingo Randolf, Iñigo Huguet, Jakub Adam, James Stevenson, Jan Alexander
+Steffens, Jan Schmidt, Jerome Laheurte, Jimmy Ohn, Joakim Johansson,
+Jochen Henneberg, Johan Bjäreholt, John-Mark Bell, John Bassett, John
+Nikolaides, Jonathan Karlsson, Jonny Lamb, Jordan Petridis, Josep Torra,
+Joshua M. Doe, Jos van Egmond, Juan Navarro, Julian Bouzas, Jun Xie,
+Junyan He, Justin Kim, Kai Kang, Kim Tae Soo, Kirill Marinushkin, Kyrylo
+Polezhaiev, Lars Petter Endresen, Linus Svensson, Louis-Francis
+Ratté-Boulianne, Lucas Stach, Luis de Bethencourt, Luz Paz, Lyon Wang,
+Maciej Wolny, Marc-André Lureau, Marc Leeman, Marco Trevisan (Treviño),
+Marcos Kintschner, Marian Mihailescu, Marinus Schraal, Mark Nauwelaerts,
+Marouen Ghodhbane, Martin Kelly, Matej Knopp, Mathieu Duponchelle,
+Matteo Valdina, Matthew Waters, Matthias Fend, memeka, Michael Drake,
+Michael Gruner, Michael Olbrich, Michael Tretter, Miguel Paris, Mike
+Wey, Mikhail Fludkov, Naveen Cherukuri, Nicola Murino, Nicolas Dufresne,
+Niels De Graef, Nirbheek Chauhan, Norbert Wesp, Ognyan Tonchev, Olivier
+Crête, Omar Akkila, Pat DeSantis, Patricia Muscalu, Patrick Radizi,
+Patrik Nilsson, Paul Kocialkowski, Per Forlin, Peter Körner, Peter
+Seiderer, Petr Kulhavy, Philippe Normand, Philippe Renon, Philipp Zabel,
+Pierre Labastie, Piotr Drąg, Roland Jon, Roman Sivriver, Roman Shpuntov,
+Rosen Penev, Russel Winder, Sam Gigliotti, Santiago Carot-Nemesio,
+Sean-Der, Sebastian Dröge, Seungha Yang, Shi Yan, Sjoerd Simons, Snir
+Sheriber, Song Bing, Soon, Thean Siew, Sreerenj Balachandran, Stefan
+Ringel, Stephane Cerveau, Stian Selnes, Suhas Nayak, Takeshi Sato,
+Thiago Santos, Thibault Saunier, Thomas Bluemel, Tianhao Liu,
+Tim-Philipp Müller, Tobias Ronge, Tomasz Andrzejak, Tomislav Tustonić,
+U. Artie Eoff, Ulf Olsson, Varunkumar Allagadapa, Víctor Guzmán, Víctor
+Manuel Jáquez Leal, Vincenzo Bono, Vineeth T M, Vivia Nikolaidou, Wang
+Fei, wangzq, Whoopie, Wim Taymans, Wind Yuan, Wonchul Lee, Xabier
+Rodriguez Calvar, Xavier Claessens, Haihao Xiang, Yacine Bandou,
+Yeongjin Jeong, Yuji Kuwabara, Zeeshan Ali,
+
+… and many others who have contributed bug reports, translations, sent
+suggestions or helped testing.
+
+
+Stable 1.16 branch
+
+After the 1.16.0 release there will be several 1.16.x bug-fix releases
+which will contain bug fixes which have been deemed suitable for a
+stable branch, but no new features or intrusive changes will be added to
+a bug-fix release usually. The 1.16.x bug-fix releases will be made from
+the git 1.16 branch, which is a stable branch.
+
+1.16.0
+
+1.16.0 was released on 19 April 2019.
+
+
+Known Issues
+
+- possibly breaking/incompatible changes to properties of wrapped
+ FFmpeg decoders and encoders (see above).
+
+- The way that GIO modules are named has changed due to upstream GLib
+ natively adding support for loading static GIO modules. This means
+ that any GStreamer application using gnutls for SSL/TLS on the
+ Android or iOS platforms (or any other setup using static libraries)
+ will fail to link looking for the g_io_module_gnutls_load_static()
+ function. The new function name is now
+ g_io_gnutls_load(gpointer data). See Android/iOS sections above for
+ further details.
+
+
+Schedule for 1.18
+
+Our next major feature release will be 1.18, and 1.17 will be the
+unstable development version leading up to the stable 1.18 release. The
+development of 1.17/1.18 will happen in the git master branch.
+
+The plan for the 1.18 development cycle is yet to be confirmed, but it
+is possible that the next cycle will be a short one in which case
+feature freeze would be perhaps around August 2019 with a new 1.18
+stable release in September.
+
+1.18 will be backwards-compatible to the stable 1.16, 1.14, 1.12, 1.10,
+1.8, 1.6, 1.4, 1.2 and 1.0 release series.
+
+------------------------------------------------------------------------
+
+_These release notes have been prepared by Tim-Philipp Müller with_
+_contributions from Sebastian Dröge, Guillaume Desmottes, Matthew
+Waters, _ _Thibault Saunier, and Víctor Manuel Jáquez Leal._
+
+_License: CC BY-SA 4.0_
projects / platform / upstream / gst-plugins-good.git / blobdiff