-This is GStreamer Base Plug-ins 0.10.25, "Standard disclaimers apply"
-
-Changes since 0.10.24:
-
- * Add per-stream volume controls
- * Theora 1.0 and Y444 and Y42B format support
- * Improve audio capture timing
- * GObject introspection support
- * Improve audio output startup
- * RTSP improvements
- * Use pango-cairo instead of pangoft2
- * Allow cdda://(device#)?track URI scheme in cddabasesrc
- * Support interlaced content in videoscale and ffmpegcolorspacee
- * Many other bug fixes and improvements
-
-Bugs fixed since 0.10.24:
-
- * 595401 : gobject assertion and null access to volume instance in playbin
- * 563828 : [decodebin2] Complains about loops in the graph when demuxer output requires another demuxer
- * 591677 : Easy codec installation is not working
- * 588523 : smarter sink selection in playbin2
- * 590146 : adder regressions
- * 321532 : [cddabasesrc] Support device setting in cdda:// URI
- * 340887 : add pangocairo textoverlay plugin.
- * 397419 : [oggdemux] ogm video with subtitles stuck on first frame
- * 556537 : [PATCH] typefind: more flexible MPEG4 start code recognition
- * 559049 : gstcheck.c:76:F:general:test_state_changes_* failure: GST_IS_CLOCK(clock) assertion fails
- * 567660 : [API] need a stream volume interface for sinks that do volume control
- * 567928 : Make videorate work with a live source
- * 571610 : [playbin] Scale of volume property is not documented
- * 583255 : [playbin2] deadlock when disabling visualisations
- * 586180 : RTSP improvements
- * 588717 : [oggmux] gst_caps_unref() warning if not linked downstream
- * 588761 : [videoscale] Needs special support for interlaced content
- * 588915 : audioresample's output offset counter's initialization could maybe be improved
- * 589095 : [appsrc] clarify documentation on caps and linkage
- * 589574 : [typefind] incorrect sdp file detection
- * 590243 : [videoscale] Claims to support MAX width/height
- * 590425 : Slaved alsasrc clock with slave-method=re-timestamp not usable for RTP audio
- * 590856 : [decodebin2] triggers assertion failure on NULL caps
- * 591207 : totem does display the following subtitle srt file.
- * 591357 : gst-plugins-base git won't build due to warning in gstrtspconnection.c
- * 591577 : [playbin2] Incorrect error message string
- * 591664 : [playbin2] after seeking, srt subtitles don't resync correctly
- * 591934 : timestamp drift in audioresample
- * 592544 : Remove regex.h check
- * 592657 : [appsink] Blocks after entering on pause state
- * 592864 : deadlocks from recent inputselector/streamselector change
- * 592884 : [playbin2] g_object_get increases refcount by 2 and therefore leaves memleak
- * 593035 : gdp doesn't preserve fields of the buffers put into the caps' streamheader
- * 593284 : basertppayloader takes time in instance init
- * 594020 : Totem don't play videos from ssh remote host
- * 594094 : Playback Error playing Midi file
- * 594136 : [alsasink] Regression from 0.10.23 -- element reuse doesn't work
- * 594165 : [theoraenc] Implement support for new formats
- * 594256 : improved slave-skew resynch mechanism
- * 594258 : missing break in rtcpbuffer
- * 594275 : Add cast to navigation to fix compiler warning
- * 594623 : Expose playsink as a fully-fledged element
- * 594732 : parse error
- * 594757 : build fails due to warning in gstbasertppayload.c
- * 594993 : [introspection] pkg-config file madness
- * 594994 : [streamvolume] Add get_type function to the documentation
- * 595454 : [cddabasesrc] uri format change breaks rhythmbox
- * 545807 : [baseaudiosink] audible crack when starting the pipeline
-
-API added since 0.10.24:
-
- * gst_rtsp_connection_create_from_fd()
- * gst_rtsp_connection_set_http_mode()
- * gst_rtsp_watch_write_data()
- * gst_rtsp_watch_send_message()
- * GstBaseRTPPayload::perfect-rtptime
- * GstBaseRTPAudioPayload::gst_base_rtp_audio_payload_flush()
- * GstVideoSinkClass::show_frame()
- * GstVideoSink:show-preroll-frame
- * GST_MIXER_TRACK_READONLY
- * GST_MIXER_TRACK_WRITEONLY
- * GstStreamVolume interface
-
-Changes since 0.10.23:
-
- * Recognise Kate subpicture subtitles
- * Support progressive download in playbin2
- * GIO improvements
- * Add buffer-list support in appsink
- * Add gaussian-noise mode to audiotestsrc
- * bump cdparanoia req to 0.10.2 and improve caching
- * Improve audio source base class
- * Add frame-by-frame stepping and examples
- * Extend stream-probing in decodebin2
- * Many RTSP improvements
- * support for PGS subpictures
- * adder improvements
- * Add Y444, v210, v216 formats
- * implement preset interface in vorbisenc, theoraenc, oggmux
- * Improve libvisual visualisation timestamp tracking
- * playbin2 enhancements: custom audiosink, subpictures, cdda
- * Improvements in textrender
- * Support raw YUV 4:2:2 and SIREN in RIFF
- * Add 4:2:2 and 4:4:4 support to theoradec
- * Many other bug-fixes and improvements
-
-Bugs fixed since 0.10.23:
-
- * 510417 : [gio] make non-experimental
- * 513373 : [PATCH] [gstvorbistag] Preserve cover art in Ogg/Vorbis tags
- * 529300 : [giosink] [PATCH] Allow overwrite
- * 531035 : [cdparanoia] Should depend on LGPL'd version of the libra...
- * 567997 : [patch] add allow-pull-scheduling property to audio sinks
- * 576552 : [subparse] post GST_TAG_SUBTITLE_CODEC tags
- * 577637 : [playbin2] expose temp-location property
- * 579692 : mp3_type_find is over-optimistic
- * 580318 : [tagdemux] drops tag events from upstream
- * 581460 : [baseaudiosrc] Reusing audio source leads to null timesta...
- * 581571 : ARGB and alignment added to textrender
- * 582021 : autogen: libtoolize must be called before aclocal
- * 582749 : uridecodebin caps property not implemented yet
- * 582819 : multifdsink: add num-fds property
- * 583867 : gdpdepay + identity cause failed assertions
- * 584020 : [playbin2] inadvertently resets configured audio/video sinks
- * 584686 : [playbin2] Need {audio,video,text}-tags-changed signals
- * 585197 : [subparse] fails to detect subrip subtitles with fewer th...
- * 585758 : Remove deprecated GTK+ symbols
- * 585970 : gst_audioringbuffer_get_type is not thread safe
- * 585994 : gst-rtsp-message doesn't support " Timestamp " filed
- * 586331 : [cdparanoia] expose cd cache size parameter
- * 586356 : [playbin2] use private copy of input-selector as long as ...
- * 586519 : white Gaussian noise would be useful in audiotestsrc
- * 587080 : rtsp fails to compile - doesn't see some ws2tcpip functions
- * 587278 : Support for GstBufferList in appsink
- * 587676 : Call tzset() before localtime_r(), in e.g. gst-plugins-ba...
- * 587695 : Patches to add stream-status messages audio elements
- * 587896 : " No stream given yet " error from giostreamsrc
- * 587980 : gstchannelmix.c: protect debug code with GST_DISABLE_GST_...
- * 588078 : [playbin2] Fails to go to READY again after an error
- * 588205 : Pipeline with giostreamsrc will not enter playing state
- * 588550 : build failure in git, missing gstinterfaces-0.10
- * 588551 : queue2: download buffering fixes
- * 588724 : [vorbisdec] empty encoder string causes GStreamer
- * 588746 : [audiotestsrc] Make sure tags are properly serialized in ...
- * 588747 : [adder] Serialize incoming in-band events (tags) in the d...
- * 588748 : [adder] Check dataflow consistency in unit tests
- * 589075 : [playbin2] changing volume doesn't work after stream rest...
- * 589581 : typefinder: recognise more Kate subtitle categories
- * 589622 : Cannot use both playbin and input-selector
- * 589663 : gstreamer asserts in gstaudiofilter
- * 589797 : alsasrc does not set GstAlsaSrc- > handle to NULL after snd...
- * 590470 : [typefinding] certain flac-in-ogg files not detected any ...
- * 536313 : [cdda] Remove sha1 copy once we depend on glib-2.16
- * 579642 : [oggdemux] handle broken ogg/vorbis files better
- * 582528 : playbin2 Audio CD playback broken since
- * 583318 : Assertion from within playbin2
- * 585079 : undefined references to gst_adapter_* functions in schro
- * 585708 : [adder] Wrong handling of flushing seeks
- * 588218 : Siren in .wav support
- * 586920 : rtsp: needs < netinet/in.h > on FreeBSD
-
-API added since 0.10.23:
-
- * GstNetAddress::gst_netaddress_to_string()
- * Add gst_rtsp_watch_queue_data()
- * playbin2: Add {audio,video,text}-tags-changed signals
- * Add gst_color_balance_get_balance_type()
- * Add gst_mixer_get_mixer_type()
-
-Changes since 0.10.22:
-
- * New navigation API to support DVD playback
- * playbin2 improvements
- * RTSP extensions to allow extra headers and options
- * Replace audioresampler with speexresample based code
- * Support interlacing flags in the gstvideo library
- * Support new RIFF formats
- * Improve typefinding
- * Support more frame formats in videoscale
- * Many other bug-fixes and improvements
-
-Bugs fixed since 0.10.22:
-
- * 577637 : [playbin2] expose temp-location property
- * 580120 : [playbin2] unit test fails
- * 478512 : [alsamixer] volume control slider not working
- * 574962 : rhythmbox crash in flac_type_find
- * 564139 : Documentation of TCP plugins
- * 577436 : xvimagesink should use xcontext- > depth and not count bits...
- * 350311 : [playbin2] support for subpicture subtitles
- * 378094 : Enable pango elements to handle UYVY
- * 543591 : Gnonlin can not play theora streams
- * 553295 : [riff] fuzzed AVI file causes segfault
- * 565105 : Gstreamer does not change from READY back to PAUSED in sa...
- * 565777 : [riff] unrecognised video fourcc 0x10000002 for mpeg2 in avi
- * 566661 : [typefind] Fall back to file extension using uri query
- * 567255 : [riff] doesn't detect codec_id 0x706d as AAC (amongst other)
- * 567636 : [pbutils] Missing plugins code shouldn't ask for the same...
- * 567740 : bogus warning in decodebin2?
- * 568482 : linking problems in gst-plugins-base
- * 569655 : [ffmpegcolorspace] Add UYVY422 to GRAY8 conversion function
- * 570142 : Documentation is broken for uridecodebin
- * 570356 : aac typefinder failure
- * 570768 : [ximagesink] wrong mouse pointer position if output windo...
- * 570832 : Add flags to enhance mixer interfaces
- * 571009 : [tagdemux] WMA file with id3v2 tag causes assertion to fail
- * 571147 : [ffmpegcolorspace/videotestsrc] Add support for packed/pl...
- * 572577 : [playbin2] deadlock on shutdown
- * 572872 : [ffmpegcolorspace] Add YVYU colorspace
- * 572993 : [subparse] broken libregex dependency on Windows
- * 573165 : Generate additional export files for gstreamer app plugin
- * 573528 : Wrong format modifier in gstgiobasesink.c
- * 573529 : In gstrtspconnection.c some functions are called with wro...
- * 574293 : [decodebin2] deadlock on shutdown
- * 574319 : Missing HAVE_PROCESS_H in win32/common/config.h
- * 574447 : gstadder.c: line 904: error C2036: 'gpointer' : unknown size
- * 574939 : [typefinding] flac typefinder mis-typefinds PDFs as flac ...
- * 575550 : srt subtitle file keeps playbin2 from playing
- * 575638 : kissfft copyright
- * 575649 : [oggdemux] duration query in time format returns true wit...
- * 576019 : On Windows queue2 can't write files longer than 2-4 GiB, ...
- * 576142 : [vorbisenc] Non-header output buffers have NULL caps
- * 576180 : [playbin2] Uses unref'd audiosink volume if using gconfau...
- * 576586 : [alsamixer] gnome-sound-properties freeze
- * 577054 : [videoscale] Not valgrind clean
- * 577709 : Review new navigation API
- * 577827 : [appsink] Have appsink new_buffer-callback return GstFlow...
- * 578583 : [PATCH] multifdsink doesn't handle sync-method=latest-key...
- * 578656 : Implement upstream GstForceKeyUnit events in theoraenc
- * 579129 : pkgconfig: appsrc/appsink can not be linked to uninstalled
- * 579130 : app: expose trivial type macros
- * 579192 : gst_rtcp_packet_get_type should not assert on packet content
- * 579203 : baseaudiosink: unparenting the ringbuffer in NULL causes ...
- * 579267 : [rtspconnection] g_async_queue_new_full() is GLib-2.16 AP...
- * 579463 : [cddabasesrc] [cdparanoiasrc] no longer emits discid
- * 579668 : audioresample fails to build with --disable-gst-debug
- * 579734 : [playbin] raw_decoding_mode seems to be set unconditionally
- * 579912 : [decodebin2] multiqueue is too small in time (interleave ...
- * 580470 : [audioresample] causes pipelines to go out of sync and be...
- * 580952 : [audioresample] bad quality/pops compared to plughw
- * 581727 : [playbin2] make playsink go to PAUSED async
- * 569682 : playbin2 leaks request pad from input selector
- * 580020 : [vorbisenc] causes buffers to be out of segment if new se...
- * 562794 : rtspsrc fails to create a socket on Win32 sometimes.
- * 567396 : playbin2: DECODE_BIN_LOCK occasionally called twice withi...
- * 567982 : " queued_bytes " field isn't updated while flushing the que...
- * 571299 : [appsink] Handoff callback API
- * 574443 : rtsp win32 - forgotten variable
- * 574516 : [typefind] add typefinder for photoshop .psd files
- * 574964 : gst_app_src_end_of_stream(), mutex on error return
- * 575256 : rtspsrc fails to resolve hostnames
- * 575588 : decodebin2 deadlock
- * 576187 : [playbin2] Stalls video sink when disabling subtitles in ...
- * 576188 : [playbin2] Reusing a playbin2 instance with visualization...
- * 576190 : [playbin2] Deadlock when reusing playbin2 after an error
- * 577288 : " Internal playbin error " when seeking to the end of files
- * 577610 : RTCP feedback messages support in GstRTCPPacket
- * 577794 : [playbin2] leaks elements set through properties
- * 578118 : [multifdsink] add option to not resend the streamheader w...
- * 578506 : Pipeline with alsasrc and alsasink cannot change state ba...
- * 578942 : Missing RTSP headers related to Windows Media extension.
- * 580271 : videorate: fails to clear discont flag on duplicated buffers
- * 580649 : uridecodebin: bug on documentation published in website
-
-API added since 0.10.22:
-
- * GstRTSP::gst_rtsp_options_as_text()
- * GstRTSPMessage::gst_rtsp_message_take_header()
- * GstRTSPRange::gst_rtsp_range_to_string()
- * New Navigation interface commands, queries and messages
- * gst_rtsp_channel_new()
- * gst_rtsp_channel_unref()
- * gst_rtsp_channel_attach()
- * gst_rtsp_channel_queue_message()
- * gst_rtsp_connection_accept()
- * GstAppSink::gst_app_sink_set_callbacks()
- * GST_VIDEO_FORMAT_YVYU,GST_VIDEO_BUFFER_TFF,GST_VIDEO_BUFFER_RFF,GST_VIDEO_BUFFER_ONEFIELD
- * GST_MIXER_FLAG_HAS_WHITELIST,GST_MIXER_FLAG_GROUPING,GST_MIXER_TRACK_NO_RECORD,GST_MIXER_TRACK_NO_MUTE,GST_MIXER_TRACK_WHITELIST
- * GstAppSrc::emit-signals
- * GstAppSrc::gst_app_src_set_emit_signals()
- * GstAppSrc::gst_app_src_get_emit_signals()
- * GstAppSrc::gst_app_src_set_callbacks()
- * RTSP::gst_rtsp_connection_get_url()
- * GstRTSPLowerTrans::GST_RTSP_LOWER_TRANS_HTTP
- * RTSP:gst_rtsp_connection_set_tunneled()
- * RTSP:gst_rtsp_connection_is_tunneled()
- * RTSP::gst_rtsp_connection_set_ip()
- * RTSP::gst_rtsp_connection_get_tunnelid()
- * RTSP::gst_rtsp_connection_do_tunnel()
- * RTSP::gst_rtsp_watch_reset()
-
-IMPORTANT NOTES
-
-1) Please note that decodebin2 and playbin2 API included in this release is
-still considered unstable and WILL change in future releases. At this stage,
-only developers or early adopters should consider using decodebin2 or playbin2
-API embodied in their signals and properties.
-
-Changes since 0.10.21:
-
- * Require gettext 0.17
- * Replace audioresample with speexresample from -bad
- * Support new formats in RIFF: uncompressed RGB, WMA lossless, VP6
- * Move libgstapp and elements from -bad
- * Support color-key setting and probing for Xv properties
- * Improve typefinding for various formats
- * Extend audio sinks for pull-mode operation
- * Support for more subtitle formats
- * More development on decode2bin and playbin2
- * RTP and SDP fixes
- * Many bug fixes and improvements
-
-Bugs fixed since 0.10.21:
-
- * 562163 : theoraenc likely ignoring segments
- * 562258 : rtspsrc element takes long time to error out if the addre...
- * 561789 : [volume] deadlocks with a controller attached
- * 554533 : [xvimagesink] allow setting colorkey if possible
- * 567511 : colorkey in xvimagesink gets reset when element is reused
- * 116051 : libresample doesn't handle > factor of 2 rate conversion
- * 346218 : [audioresample] doesn't do anti aliasing
- * 385061 : [audioresample?] investigate high CPU usage
- * 456788 : [subparse] can't handle UTF-16 charset encoded subtitle.
- * 525807 : [vorbisenc] vorbisenc has problems with a gnlsource that ...
- * 546955 : gstoggmux EOS handling issue
- * 549417 : [audioresample] unit test fails on 64bit linux
- * 549510 : audioresample doesn't negotiate ideal caps
- * 552237 : UTF-16 srt confuses gstreamer, misdetected as mp3
- * 552559 : Implementation of SLAVE_SKEW in baseaudiosrc
- * 552569 : audioresample producing strange sized buffers
- * 552801 : audioconvert can overflow with big audio buffers
- * 554879 : Add ability to specify format for date/time display in Gs...
- * 555257 : Doesn't display srt subtitles saved with BOM
- * 555319 : add FFV1 fourcc to riff-media
- * 555607 : subrip subtitles typefind too strict
- * 555699 : [PATCH] theoradec: prefer container's pixel aspect ratio ...
- * 556025 : build failure in tests/icles
- * 556066 : Last byte of FLAC image buffer chopped off
- * 557365 : subparse check fails
- * 558124 : [PLUGIN-MOVE] Move speexresample as audioresample2 to -base
- * 559111 : ALSA sink hangs on USB audio device unplug while playing
- * 559478 : does not play windows media streams correctly
- * 559567 : `gst_base_audio_sink_sync_latency' should call `gst_base_...
- * 561436 : videorate element add image/jpeg to caps template
- * 561734 : playbin2 additions
- * 561780 : Playbin2 should work without volume too
- * 561924 : oggdemux hangs when given corrupt input via non-seekable ...
- * 562270 : build without gdk fails
- * 563143 : ximagesink/xvimagesink : _alloc_buffer returns non-clean ...
- * 563174 : Implement gst_rtcp_packet_remove
- * 563508 : [rgvolume] Unit test fails with passthrough assertions
- * 563718 : Theora check out of date
- * 563904 : GNOME Goal: Clean up GLib and GTK+ includes
-
-API added since 0.10.21:
-
- * clockoverlay::time-format
- * GstRingBuffer:gst_ring_buffer_activate()
- * GstRingBuffer:gst_ring_buffer_is_active()
- * GstRingBuffer:gst_ring_buffer_convert()
- * Add GST_TYPE_BASE_AUDIO_(SRC|SINK)_SLAVE_METHOD to the public API
- * gst_netaddress_get_address_bytes()
- * gst_netaddress_set_address_bytes()
-
-Changes since 0.10.20:
-
- * Continue playbin2 development
- * Ogg improvements - CELT support, skeleton fixes
- * DVD subpicture support
- * Improved audio dithering random number generator
- * xvimagesink/ximagesink fixes
- * Vorbis encoding and decoding fixes
- * Recognise Kate subtitle streams
- * Many bug-fixes and enhancements
-
-Bugs fixed since 0.10.20:
-
- * 537380 : [gnomevfssrc] Doesn't handle short reads properly
- * 538656 : xvimagesink support for autofill/colorkey property
- * 540334 : Build fails without X in tests/examples/seek
- * 528299 : Multiple GstMixerTracks with the same label cause problem...
- * 530068 : Ogg Streams with Skeleton and Granulepos > 0 do not work(...
- * 537009 : playbin2 silly typo breaks signals
- * 537045 : decodebin2 sometimes emits 'drained' multiple times
- * 537599 : [oggdemux] skeleton streams not skipped in ogg
- * 537889 : [xvimagesink] colorbalance is bad
- * 538232 : vorbisenc/vorbisdec don't work with a live source
- * 538663 : gdppay memleak in gst_gdp_pay_reset
- * 540215 : decodebin does not insert a queue for raw data type
- * 540351 : [avidemux] Doesn't know about Duck DK4 ADPCM
- * 540497 : ffmpegcolorspace is returning wrong size
- * 541358 : cross mingw32 gcc: getaddrinfo is not in ws2_32.dll befor...
- * 544306 : rtspsrc debug=1 segfaults with some libc
- * 548898 : GStreamer-CRITICAL errors on seeking beyond stream borders
- * 548913 : vorbisenc being picky about rounding errors in timestamps
- * 549062 : Video devices aren't updated on subsequent probing.
- * 549814 : [typefind] add application/pdf typefinder
- * 550582 : [oggdemux] KATE streams not recognised
- * 550638 : [typefind] Recognize some jpeg2k file types
- * 550656 : recognize TrueSpeech in wavparse
- * 550729 : gst-plugins-base won't compile with " -pedantic " option
- * 552960 : tagdemux asserts and aborts on truncated files
- * 553244 : theoraparse doesn't work at all (throws criticals and ass...
-
-API added since 0.10.20:
-
- * Add "index" property to GstMixerTrack to differantiate between
- multiple mixer tracks with the same label.
-
-Changes since 0.10.19:
-
- * RTP improvements
- * Support digest auth for RTSP
- * Additional documentation
- * Support DSCP QoS in multifdsink
- * Add NV12/NV21 video buffer layouts
- * Video scaling now bilinear by default
- * Support more than 8 channels in audio conversions
- * Channel mapping fixes for audioconvert
- * Improve tmplayer and sami subtitle support
- * Support 1x1 pixel buffers for videoscale
- * Typefinding improvements for MPEG2, musepack
- * Ogg/Dirac mapping updated in oggmux
- * Fixes in ogg demuxing
- * audiosink synchronisation and slaving fixes
- * Support muting of the audio in playbin by selecting -1 as the audio stream
- * Work done on playbin2 and uridecodebin
- * Improvements in the experimental GIO plugin
- * decodebin fixes
- * Handle GAP buffers in some places
- * Various other leak and bug-fixes
-
-Bugs fixed since 0.10.20:
-
- * 526794 : [giosrc] totem doesn't work with some gvfs backends
- * 510417 : [PLUGIN-MOVE] Move gio to gst-plugins-base
- * 509125 : crash in CD Player: - playing CD - lowering/...
- * 517813 : [audioconvert] make gap aware
- * 302798 : [playbin] add mute property
- * 342294 : Setting playbin property current-audio=-1 also stops the ...
- * 398033 : [audioconvert] support more than 8 channels
- * 419351 : [avi/a52dec] AV synchronization problems
- * 467911 : [subparse] sami parser update
- * 469933 : multifdsink IPv6 and diffserv TOS/TC markup
- * 506659 : [textoverlay] rendering error when using non-standard widths
- * 512333 : [gstvorbistag] Retrieve Ogg/Vorbis cover art as image met...
- * 512382 : [playbin] race condition when pausing/playing multiple in...
- * 518037 : pbutils-enumtypes.c is not included in win32/vs6/libgstpb...
- * 521761 : gstaudioclock frozen the clock value until reaches latest...
- * 522401 : gdpdepay doesn't validate payload CRCs
- * 523993 : playbin2 blocks after a while when listening to a radio s...
- * 524724 : [PATCH] [baseaudiosrc] buffer-time and latency-time do no...
- * 525665 : Crash on Ogg/Vorbis with chain=NULL
- * 525915 : [streamheader] Unit test fails with " gst_adapter_peek: as...
- * 526173 : [typefinding] fails to detect mpeg video stream whereas m...
- * 529018 : gst_ogm_parse_stream_header creates fraction value with w...
- * 529500 : [videotestsrc] support for NV12 and NV21
- * 529546 : [Playbin] Memory leak in streaminfo handling
- * 530068 : Ogg Streams with Skeleton and Granulepos > 0 do not work(...
- * 530531 : [typefinding] bad read in mpeg_video_stream_type_find
- * 530719 : gst_video_calculate_display_ratio fails when playing Ogg ...
- * 530962 : [subparse] parses only every second line of TMPlayer subt...
- * 532454 : [NV12/NV21] videotestsrc and ffmpegcolorspace don't play ...
- * 533087 : GstRTSPTransport kept opaque in docs
- * 533817 : [audioconvert] Can't use default 7 channel layout / only ...
- * 534071 : Gdppay memleak
- * 534331 : race in decodebin when changing states while the internal...
- * 535356 : vorbisdec doesn't support 8 channels
- * 536475 : gdppay memleak and possible crash
- * 536521 : Refcounting errors in playbin
- * 536874 : Build failure on windows
- * 532166 : [ffmpegcolorspace] support NV12 format
- * 533617 : [audioconvert] Produces silence when converting 1/2 chann...
- * 536848 : [giosrc] Doesn't handle short reads properly
- * 536849 : [giosrc] Very slow doing any playback
- * 518082 : [alsamixer] playback volumes overwritten by capture volum...
- * 435633 : [PATCH] videorate not (fully) segment aware; causes frame...
- * 532364 : tcpclientsrc broken in 0.10.19
- * 533075 : gst_rtp_buffer_compare_seqnum doesn't do what it says
- * 533265 : [cddabasesrc] Sound Juicer cut a sector when ripping a track
-
-API additions since 0.10.20:
-
- * decodebin2::sink-caps property
- * giosrc::file property
- * giosink::file property
- * gst_base_audio_src_set_slave_method()
- * gst_base_audio_src_get_slave_method()
- * GstAudioClock::gst_audio_clock_reset()
- * GstBaseAudioSrc:actual-buffer-time property
- * GstBaseAudioSrc:actual-latency-time property
- * gst_audio_check_channel_positions()
- * add gst_tag_image_data_to_image_buffer()
- * add gst_tag_list_add_id3_image()
- * add GST_TAG_IMAGE_TYPE_NONE enum value
-
-Changes since 0.10.18:
-
- * Handle EAGAIN when polling sockets in rtspconnection
-
-Changes since 0.10.17:
-
- * Experimental GIO plugin
- * Continued playbin2 development
- * RTP fixes
- * Better network element support on Windows
- * Various other bug-fixes and improvements
-
-Bugs fixed since 0.10.17:
-
- * 509637 : [API] [basertpaudiopayload] add _set_samplebits_options()
- * 510229 : [gnomevfssrc] HTTPS support
- * 511478 : [rtpbuffer] add gst_rtp_buffer_set_extension_data function
- * 511810 : [RTSP] Uses MT-unsafe gmtime() function
- * 512899 : [alsa] gstalsasink.c:527: warning: 'snd_pcm_sw_params_set...
- * 513167 : Fix compiler warning due to disabled signals in mixertrac...
- * 514307 : [playbin] warning in nautilus, volume element can't be cr...
- * 514623 : Ogg Theora video slow
- * 514937 : Correct initialization of hints in is_multicast_address()
- * 515654 : xvimagesink doesn't build with --disable-xshm
- * 516246 : [alsasink] handle negative delay from snd_pcm_delay
- * 517420 : typefind: add h264 elementary stream discovery
- * 517991 : problems with configure file depending on GCC compiler
- * 518039 : libgstrtsp MSVC 6.0 compile error
- * 518162 : [subparse] handle italic text starting with " / " with Micr...
- * 518940 : [playbin2] make _get_*_tags() match vfuncs prototype in c...
- * 519906 : [API] add GstMixerOptions::get_values vfunc
- * 519916 : [API] add mixer-changed and options-list-changed messages
- * 520523 : [API] Unreviewed changes to ringbuffer API
- * 521743 : libgstnetbuffer.def exports not up to date
- * 522625 : [video] gst_video_format_parse_caps() broken for RGBA for...
- * 523054 : gstbasesrc crashes when called from typefind helpers
- * 511825 : [RTSP] compiler warning on FreeBSD
- * 520300 : [alsasrc] provide-clock=false messes up buffer durations
-
-API added since 0.10.17:
-
- * GstRTPBuffer:gst_rtp_buffer_set_extension_data()
- * add GST_VIDEO_FORMAT_Y41B and GST_VIDEO_FORMAT_Y42B.
- * add GstMixerOptions::get_values vfunc (#519906)
- * add gst_mixer_options_list_changed(), gst_mixer_mixer_changed() and
- gst_mixer_message_parse_options_list_changed(). Fixes #519916.
- * gst_base_rtp_audio_payload_set_samplebits_options()
- * GstNetBuffer::gst_netaddress_equal
-
-Changes since 0.10.16:
-
- * Work-around ABI breakage due to unfortunate use of the
- GST_DISABLE_DEPRECATED macro
- * Export 2 missing functions needed for bindings in the win32 build
- * Initialise the GstRingBuffer GType from a thread-safe context
-
-Bugs fixed since 0.10.16:
-
- * 511825 : [RTSP] compiler warning on FreeBSD
- * 513018 : crash in Volume Control: I typed my password at t...
- * 512334 : g_critical() when using GstAudioFilter & GST_DEBUG
-
-Changes since 0.10.15:
-
- * Handle newer Theora granule-pos semantics
- * Introducing first alpha version playbin2 - the upcoming successor to
- playbin
- * Fixes in playbin handling of stream-switching
- * New API for uniform handling of raw-video format buffers.
- * Improvements for RTSP/RTP handling
- * RIFF lib additions for VC-1 and AVC1 fourccs
- * Many other bug-fixes and improvements
-
-Bugs fixed since 0.10.15:
-
- * 506132 : Review of changes in video/video.h
- * 320984 : [oggdemux] cannot handle multiple chains
- * 373011 : [playbin] throws error when switching off subtitles
- * 436756 : Intermittent crashes in Pidgin in audioclock g_type_class...
- * 462740 : [streamselector] patch to improve default stream selection
- * 486840 : [alsamixer] use _all variants when setting the mixer
- * 497964 : theoraenc test fails
- * 498228 : gst-plugins-base-0.10.15 does not compile on FreeBSD (Gen...
- * 499697 : Provide better pkg-config files
- * 502497 : [subparse] SubRip subtitles starting from 0 not recognised
- * 503440 : The control sockets used by gstrtspconnection.c are never...
- * 503930 : [cdda] warning: 'eos' may be used uninitialized in this f...
- * 506928 : [alsamixer] add " PCM " as master fall back for cards that ...
- * 508138 : [decodebin] does not error out if pad activation fails
- * 509762 : missing file in win32/MANIFEST
- * 511274 : gst_rtp_buffer_get_extension_data is returning FALSE when...
- * 496731 : [PATCH] xvimagesink leaks memory if initialization fails
- * 496761 : [PATCH] RTSP message leaks memory when uninitialized
- * 500763 : SIGSEGV while playing ogg audio file
-
-API additions since 0.10.15:
-
- * New GstVideoFormat API and helper functions in libgstvideo
- * gst_base_audio_sink_set_provide_clock()
- * gst_base_audio_sink_get_provide_clock()
- * gst_base_audio_sink_set_slave_method()
- * gst_base_audio_sink_get_slave_method()
- * gst_base_audio_src_set_provide_clock()
- * gst_base_audio_src_get_provide_clock()
-
-Changes since 0.10.14:
-
- * RTP/RTSP/RTCP/SDP support improved
- * New FFT support library libgstfft, based on Kiss FFT
- * New formats supported in volume and audiotestsrc
- * Fixes in audiorate and videorate
- * Audio capture fixes
- * Playbin and decodebin fixes
- * New tagdemux base class for ID3/APE style tag readers
- * Fix a nasty crash in the X sinks on shutdown
- * New tags supported
- * Add support for multichannel WAV files.
- * Preserve channel layout information when up/down-mixing.
- * Many bug-fixes and improvements
-
-Bugs fixed since 0.10.14:
-
- * 475395 : decodebin2 leaks request-pads
- * 475451 : [decodebin2] leaks ghostpad
- * 378770 : [xvimagesink] race condition in event thread?
- * 407282 : [decodebin2] autoplug-sort signal has GList ** parameter
- * 430677 : [audioconvert] does not preserve channel positions when f...
- * 442654 : [volume] controller bypassed by default
- * 445529 : [volume] support for 24/32-bit audio/x-raw-int
- * 446766 : return code for gst_base_rtp_payload_audio_handle_event()
- * 451970 : Subparse requires HTML parser
- * 453650 : [audiobasesrc] two alsasrcs do not work in one pipeline
- * 459334 : [textoverlay] expose pango line alignment property
- * 459585 : [basertpdepayload] api without namespace
- * 460422 : [audiotestsrc] Add support for float and double output
- * 462805 : [alsa] compilation fails with gcc 4.2
- * 462979 : Add 'silent' property to GstTimeOverlay
- * 463215 : [audioconvert] compile errors
- * 464320 : [PATCH] gst-plugins-base-0.14 does not build for win32
- * 464666 : [playbin] QT trailer hangs in preroll with decodebin2
- * 464690 : Add connection-speed property to uridecodebin element
- * 465015 : [playbin] Not removed probes causes deadlocks in streamin...
- * 465028 : some warnings with mingw
- * 467667 : GST_FRAMES_TO_CLOCK_TIME() and GST_CLOCK_TIME_TO_FRAMES()...
- * 468129 : [basertpaudiopayload] event handler returns the wrong value
- * 468619 : New library gstfft: FFT library for integer and float typ...
- * 470456 : [API] add gst_missing_*_installer_detail_new()
- * 470766 : [ssaparse] line breaks in SSA subtitle parser
- * 471067 : Make the SDP code useable for generating SDP descriptions
- * 471194 : [rtpbuffer] RTP headers are wrong for win32
- * 473097 : [baseaudiosink] gstreamer-properties hangs when testing s...
- * 474384 : gstrtsp-enumtypes.c and .h needed for win32
- * 474880 : [xvimagesink] [ximagesink] leaking buffer caps reference
- * 475731 : rtspconnection is able to read incomplete messages
- * 483620 : All Rtp buffers are discarded -- gst_rtp_buffer_get_payl...
- * 484989 : memleak, not unrefed caps for gstbasertppayload.c
- * 489010 : Please change default channel order for WAVE_EXT-less .wa...
- * 491722 : [playbin] regression: crash with external subtitles
- * 492098 : [GstFFT] Broken scaling
- * 492114 : Build issues on Windows/MSVC
- * 492306 : compilation errors with MinGW
- * 492813 : Missing symbols in libgstrtp.def
- * 493986 : Build issues on Windows (missing symbols)
- * 494346 : pre-release vs6 patch
- * 496548 : Including malloc.h breaks macos build
- * 496724 : DSW file references non-existent DSP files
- * 464079 : audiotestsrc doesn't respond to conversion queries properly
- * 442065 : floatcast.h includes config.h and might break other apps
- * 466717 : gst_event_new_new_segment_full:assertion `start < = stop' ...
- * 485753 : Decodebin2 deadlocks when nulling pipeline during typefind
- * 464028 : Move connection-speed from playbin to playbasebin
-
-API added since 0.10.14:
-
- * GstTagDemux base class for simple tag demuxers
- * GstBaseAudioSrc::provide-clock property
- * gst_rtcp_ntp_to_unix()
- * gst_rtcp_unix_to_ntp()
- * gst_rtp_buffer_get_header_len()
- * gst_rtp_buffer_get_extension_data()
- * gst_rtp_buffer_compare_seqnum()
- * gst_rtp_buffer_ext_timestamp()
- * gst_rtcp_packet_sdes_copy_entry()
- * gst_install_plugins_supported()
- * gst_missing_*_installer_detail_new() convenience API
- * gst_rtsp_connection_poll()
- * GstTextOverlay::line-alignment property
-
-Changes since 0.10.13:
-
- * Audio dither and noise-shaping when reducing bit-depth
- * RTSP and SDP helper libraries added
- * Experimental buffering element "queue2" now supports pull-mode
- and file-based buffering.
- * Support for more 32-bit video pixel layouts
- * Various fixes and improvements
-
-Bugs fixed since 0.10.13:
-
- * 380625 : [x*imagesink] add 'handle-expose' property
- * 385527 : oggmux sometimes gets DELTA flag on output wrong near start
- * 402076 : videoscale 4-tap method broken for downscaling
- * 437169 : [xvimagesink] add property to disable Xv double-buffering
- * 441264 : queue2 support to do buffering on a file
- * 442553 : [v4lsrc] doesn't output segments in GST_FORMAT_TIME
- * 442557 : [videorate] doesn't handle latency queries
- * 442944 : Audiotestsrc can overflow on seeks
- * 444523 : [queue2] Pull mode support
- * 444630 : Compilation error with fsseko (from gstqueue2.c) -- unabl...
- * 445505 : [queue2] It does not work in pull mode with oggdemux
- * 446551 : [queue2] Buffering is not working properly if it is set t...
- * 446572 : [queue2] Division by zero
- * 446972 : warning when compiling gstoggdemux.c
- * 449156 : Regression in CVS for decodebin2
- * 450875 : Missing files in po/POTFILES.in
- * 451707 : [tag] UTF-8 in ID3v1 tag not correctly decoded
- * 451908 : [ffmpegcolorspace] regression: doesn't accept GST_VIDEO_C...
- * 454264 : Playbin fails to " play " image url after a movie url
- * 456656 : [API] Addition of audio buffer clipping function to gstaudio
- * 460978 : gst_audio_buffer_clip outputs warnings
- * 152864 : [PATCH] GstAlsaMixer doesn't support signals
- * 360246 : [audioconvert] Optionally apply dithering
- * 394061 : Add support for Subviewer subtitles
- * 420326 : Base payloader class has wrong property types and ranges
- * 451145 : [vorbisdec] errors out on 0-sized packets
- * 459204 : [PATCH] [playbin] gst_play_base_bin_get_streaminfo_value_...
-
-API added since 0.10.13:
-
- * RTSP and SDP libraries added
- * gst_rtsp_base64_decode_ip
- * Add buffer clipping function gst_audio_buffer_clip for raw audio
- buffers. Fixes #456656.
- * gst_mixer_get_mixer_flags
- * gst_mixer_message_parse_mute_toggled
- * gst_mixer_message_parse_record_toggled
- * gst_mixer_message_parse_volume_changed
- * gst_mixer_message_parse_option_changed
- * GstMixerMessageType
- * GstMixerFlags
-
-Changes since 0.10.12:
- * Many fixes and improvements
- * RTP and RTCP support improved
-
-Bugs fixed since 0.10.12:
-
- * 339838 : [audioconvert] support floats with non-native endianness
- * 393975 : closing x/xvimagesink window crashes gst-launch
- * 405072 : [API] add gst_tag_freeform_string_to_utf8()
- * 413799 : [subparse] add support for MPL2 format
- * 414645 : GstMixerTrack should make untranslated label available
- * 420079 : [audioconvert] Uses biased rounding which results in dist...
- * 420578 : [subparse] add more colour map in sami parser
- * 421834 : videorate breaks on dimension changes
- * 423051 : Vorbis tags of type double use locale-dependent formatting
- * 423055 : Verify ReplayGain vorbistag processing in libs/tag testsuite
- * 425455 : Decodebin2 leaks pads
- * 426250 : GstPlayBaseBin leaks streaminfo objects
- * 428187 : Rtp base depayloader class doesn't send new_segment after...
- * 431672 : gst_base_rtp_audio_payload_push() should take object of i...
- * 432362 : [ximagesink] doesn't build if XShm is not available
- * 432755 : [videorate] leaks buffer if flow != OK
- * 432984 : [baseaudiosrc] misleading warning message when dropping s...
- * 433888 : [theoradec] does not generate a perfect stream
- * 436562 : Theoradec doesn't work well with gnonlin
- * 438840 : [theoradec] does not compile with old version of libtheora
- * 440997 : [gstriff] Doesn't handle width!=depth files with audio/x-...
- * 441295 : audioconvert doesn't build on VS6
- * 442024 : regression in playbin buffering
- * 350299 : [playbin] " Internal data flow error " opening movie with s...
- * 410039 : totem crashed with SIGSEGV in new_decoded_pad_full()
- * 340842 : do latency calculation for live sources
- * 341078 : RB does not play beyond initially downloaded podcast file
- * 414496 : [id3demux, id3v2mux] Add support for GST_TAG_MUSICBRAINZ_...
-
-API additions since 0.10.12:
-
- * add gst_tag_freeform_string_to_utf8()
- * GstRTPBuffer::gst_rtp_buffer_default_clock_rate()
- * GstBaseAudioSink::slave-method property
- * add "min-ptime" property to RTP base audio payloader
- * gst_base_rtp_audio_payload_push()
- * gst_base_rtp_audio_payload_get_adapter()
- * GstMixerTrack::untranslated-label property
-
-Changes since 0.10.11:
-
- * New API for on-demand plugin installation
- * Xv thread-safety and configuration enhancements
- * decodebin2 improvements
- * Support more raw audio format conversions
- * Improvements in Ogg support
- * AudioFilter base class ported to 0.10
- * Fixes for subtitles
- * Latency/live-playback support for Alsa
- * Lots of bug fixes and improvements
-
-Bugs fixed since 0.10.11:
-
- * 398721 : No video in .ogm files with decodebin2
- * 339837 : [audioconvert] support for 64-bit float audio
- * 341524 : [decodebin] can't handle decoders with always src pads wi...
- * 352069 : Add de.po German translation
- * 363379 : [oggmux] doesn't detect EOS on all sinkpads
- * 378436 : [oggdemux] rhythmbox crash on fast clicking on rating in ...
- * 380342 : Totem does not play mp3 files when lyrics are present
- * 383195 : [cddabasesrc,basertpaudiopayload] compile errors with gcc...
- * 383198 : totem crashed to gst_xvimagesink_update_colorbalance
- * 384008 : [xvimagesink] accesses - > xwindow outside locks
- * 384060 : gst_xoverlay_set_xwindow_id() causing lockups with x(v)im...
- * 387138 : x input events processing in sinks with xoverlay interfac...
- * 390063 : Documentation typo
- * 390076 : add xv adaptor and port properties in xvimagesink element.
- * 391365 : [oggdemux] internal stream error on OggFlac
- * 392070 : [vorbis] GST_TAG_LOCATION not mapped
- * 392393 : [API] add libgstbaseutils library for missing plugins mes...
- * 396042 : mpeg4 video typefinder loops endlessly on quicktime redirect
- * 396835 : audioconvert/audioresample combination causing buffer of ...
- * 397673 : [patch] XIOError caught in x[v]imagesink.c
- * 397810 : [typefinding] .vob file: could not determine type of stream
- * 398110 : [theoraenc] GLib failed to allocate 3080991032 bytes on g...
- * 399340 : Crash in the oggdemux plugin when trying to play a specia...
- * 401029 : [playbin] rapidly changing visualisation freezes
- * 401072 : Move libgimme-codec helper functions to GStreamer
- * 402505 : visualisations don't work for some samplerates
- * 407811 : decodebin2 hang on HD clip
- * 409683 : Crash with Decodebin2
- * 410396 : not reading " DATE " tags from Flac files
- * 410963 : Fails to build with -z defs
- * 357503 : [suparse] wrong timing with microdvd subtitles
- * 393310 : [pango] localtime_r does not exist in MinGW
- * 397207 : Test failure w/ HP-UX 11.11 & native compiler
- * 399948 : [textoverlay] leaks upstream events if textpad unlinked
- * 403963 : GstAudioFilter base class broken
- * 404512 : [videoscale] floating point exception on 1x1 video
- * 405020 : [alsa] probing the device-name doesn't seem to work corre...
- * 408278 : [videorate] memory leak
- * 410772 : Crash copying a GstNetBuffer
- * 401118 : [visual] error if width not a multiple of 4
- * 405451 : [alsasink] deadlocks when disconnecting USB Sounddevice
-
-API additions since 0.10.11:
-
- * GstAudioFilter
- * GST_VIDEO_SINK_CAST()
- * gst_pb_utils_add_codec_description_to_tag_list()
- * gst_pb_utils_get_codec_description()
- * gst_pb_utils_get_source_description()
- * gst_pb_utils_get_sink_description()
- * gst_pb_utils_get_decoder_description()
- * gst_pb_utils_get_encoder_description()
- * gst_pb_utils_get_element_description()
- * gst_pb_utils_init()
- * gst_install_plugins_context_new()
- * gst_install_plugins_context_set_xid()
- * gst_install_plugins_context_free()
- * gst_install_plugins_async()
- * gst_install_plugins_sync()
- * gst_install_plugins_return_get_name()
- * gst_install_plugins_installation_in_progress()
- * gst_missing_uri_source_message_new()
- * gst_missing_uri_sink_message_new
- * gst_missing_element_message_new
- * gst_missing_decoder_message_new
- * gst_missing_encoder_message_new
- * gst_missing_plugin_message_get_installer_detail
- * gst_missing_plugin_message_get_description
- * gst_is_missing_plugin_message
-
-Bugs fixed since 0.10.10:
-
- * 360552 : [riff] [avi] extracts non-UTF8 metadata
- * 365501 : [x/xvimagesink] race condition when creating first image ...
- * 339366 : [playbin] hangs if suburi file type cannot be determined
- * 355914 : libvisual causes xvimagesink: assertion `GST_CAPS_REFCOU...
- * 363118 : gst_riff_create_video_caps() should also store variant in...
- * 363607 : xvimagesink xwindow_draw_border() slowness
- * 336301 : [playbin] can't handle RTSP source
- * 337026 : oggmux doesn't set EOS properly
- * 337031 : vorbisdec outputs too much data
- * 340049 : New BaseRTPAudioPayloader class to -base
- * 348264 : Theora encoding, Ogg muxing don't handle discontinuities
- * 354773 : xvimage assumes that XV_COLORKEY can be set in RGB888 format
- * 355917 : libvisual plugin is broken
- * 355935 : multifdsink doesn't allow setting maximums (soft, hard) i...
- * 357038 : [ffmpegcolorspace] RGBA handling broken
- * 357215 : [playbin] buffering notification not quite right yet
- * 357289 : [riff] riff parser can't detect aac audio stream
- * 357404 : [playbin] Linking can fail silently
- * 357531 : [subparse] problem if markup is not closed
- * 357577 : [playbin] regression: buffering still images broken
- * 357591 : Avoid compiler warning with uclibc and -Werror
- * 357613 : XvStopVideo in xvimagesink
- * 357800 : [libvisual] doesn't pass audio data to libvisual 0.4.0 co...
- * 359580 : tcpserversink and dataprotocol assert for multipart streams
- * 361095 : Fixes compiling with forte: warning clean up (part 3)
- * 361456 : [basertppayload] Memory leak
- * 361634 : sink- > ringbuffer NULL in BaseAudioSink's setcaps()
- * 361984 : [subparse] doesn't accept .srt file that doesn't start wi...
- * 366334 : [PATCH] Windows vs8 fixes
- * 368273 : Using the remove signal on multifdsink is not threadsafe
- * 368310 : include file gstbasertpaudiopayload.h not included for r...
- * 369482 : [typefind] MPEG system streams get recognized as mp3 files
- * 370092 : [PATCH] Decodebin v2 : Implementation
- * 377183 : regression: no eos when playing ogg vorbis files
- * 381219 : bad debugging code left in audiorate
- * 382223 : [decodebin] more delayed linking
- * 382269 : Typefind detects mpeg video clip as audio/mpeg
- * 335635 : Add an Ogg/Vorbis retagging element
- * 341681 : [textoverlay] flickering with continuously timestamped text
- * 342228 : [alsa] Recognize " Front " as a Master channel
- * 357330 : [subparse] some sami parser minor but enhanced patch
- * 357532 : [gsttag] vorbistag doesn't handle dates that include time...
- * 359237 : [typefinding] doesn't recognize XML files shorter than 25...
- * 362845 : [subparse] add support for tmplayer format
- * 357977 : [videorate] new segment start is not respected
- * 364812 : [PATCH] oggmux release pad does not remove pad
- * 364856 : pngenc stride problems
- * 372507 : Mac build fixes
-
-API added since 0.10.10:
-
- * playbin::queue-min-threshold property.
- * GstVideoOrientation interface
- * gst_base_rtp_depayload_push_ts
- * gst_base_rtp_depayload_push
- * Add dropped_buffers to multifdsink's get-stats GValueArray
- * gst_ring_buffer_commit_full
-
-Changes since 0.10.9:
-
- * New elements: gdppay, gdpdepay
-
-Bugs fixed since 0.10.9:
-
- * 343787 : The adder cannot handle when multiple elements tries to l...
- * 336075 : ALSA emu10k1 mixer tracks are wrongly classified as playb...
- * 349105 : crash with playbin and resizing screen
- * 342494 : [v4l] Query " device-name " even if device is not open
- * 342680 : [adder] seeking with multiple ogg files fails to work
- * 345188 : [alsa] can't handle more than 8 channels
- * 347091 : converting vorbis comments to GstTagLists is lossy
- * 348157 : Changed " Change Device " menu behaviour in gnome-volume-co...
- * 348916 : [typefind] add multipart/x-mixed-replace typefinder
- * 350157 : [riff] riff parser can't detect dts audio stream
- * 350655 : [oggdemux] should process seeking queries
- * 350900 : [adder] should not clamp floating point values
- * 351426 : API: add gst_tag_parse_extended_comment
- * 351502 : g_value_set_string leaks
- * 351742 : [vorbisenc] discontinuity detection too sensitive, might ...
- * 353658 : [videotestsrc] doesn't round strides correctly for YVYU
- * 354594 : multifdsink doesn't work reliably with sync-method = 'nex...
- * 351790 : [ogmparse] crash parsing video stream on x86-64
- * 140139 : [avidemux] can't play broken avi with ogg (not vorbis) au...
- * 347783 : [PLUGIN-MOVE] GDP elements should be moved
- * 347918 : Internal data flow error in udpsrc
- * 349656 : jitterbuffer in GstBaseRtp fails to handle rtp seqnum rol...
- * 350784 : element alsamixer doesn't respect asoundrc
- * 351308 : [netbuffer] build fails with gkt-doc critical warnings
- * 353234 : audiorate preserves DISCONT on buffers
- * 353912 : Add cmml caps to oggmux
-
-API added since 0.10.9:
-
- * gst_rtp_buffer_get_payload_subbuffer()
- * gst_tag_parse_extended_comment()
- * GstPlayBin::connection-speed
- * GstTheoraParse::synchronization-points
- * GST_AUDIO_CHANNEL_POSITION_NONE
-
-Changes since 0.10.8:
-
- * Parallel installability with 0.8.x series
- * Threadsafe design and API
- * Subtitle fixes
- * Support for images in tags
- * Playback improvements
- * Gnomevfssrc now supports burn:// uris
- * Videoscale now supports more RGBA formats
- * Multifdsink improvements
- * Testsuite can now generate coverage information
-
-Bugs fixed since 0.10.8:
-
- * 347296 : Problems with clocks on alsasrc hangs the application
- * 347295 : [vorbisdec] Pushes before being initialized
- * 329798 : [playbin] doesn't always give correct error message for m...
- * 342085 : [alsasink] doesn't set buffer-time correctly
- * 342789 : [audioresample] doesn't clear state when stopped, causing...
- * 343303 : [subparse] workaround for bad entities in sami parser
- * 343385 : [gnomevfs] add support for burn:// URIs
- * 343500 : [riff] gst_riff_parse_strf_vids() can't parse extra data.
- * 343699 : oggmux leaks
- * 344503 : [subparse] parse font face property in sami parser.
- * 345131 : [PATCH] videoscale support for 32-bit RGB-formats
- * 345206 : [textoverlay] crash with non-UTF8 input
- * 345225 : [theoradec] Clipping for exact seeking
- * 345641 : [API] [libgsttag] add enums for image tag type
- * 345879 : [riff] won't play a .wmv file with WMVA video stream
- * 346581 : [typefinding] recognise text/html
- * 347221 : [audioconvert] channel remapping does not work right
- * 347304 : Massive leaks with xvimagesink
- * 346527 : alsasrc get_range does not respect requested size
-
-Changes since 0.10.7:
-
- * alsasink probing fixes
- * xvimagesink error reporting fixes
- * subtitle fixes
- * adder fixes
- * vorbis multichannel fixes
- * multifdsink streamheader fixes
-
-Bugs fixed since 0.10.7:
-
- * 169936 : [subparse] support for SAMI subtitles
- * 315312 : Gstreamer Xv uses RGB instead of YUV.
- * 334002 : video4linux shouldn't depend on X in configure script
- * 336881 : [libvisual] additional support for libvisual-0.4
- * 337544 : [xvimagesink] Internal Error when image is too large
- * 339520 : [subparse] add " encoding " property
- * 340909 : [alsasink] can't enable spdif output
- * 341542 : some users have an assertion failed: (GST_VIDEO_SINK_WIDT...
- * 341562 : audioconvert doesn't list formats in order of preference
- * 341696 : audioconvert crashes if converting from a format with no ...
- * 341719 : bisection algorithm in ogg doesn't bisect in some cases
- * 341732 : [alsasink] doesn't query supported sample rates
- * 341873 : [alsasink] minor memory leak, uses unprotected static var...
- * 342143 : [subparse] sami parser needs to escape characters
- * 342181 : [alsa] add property probe interface to alsasink and alsasrc
- * 342268 : [playbin] add 'subtitle-encoding' property
- * 342345 : [riff] Elephant's Dream AVI does not play, JUNK chunk bef...
- * 342566 : Building without GTK+ fails
- * 343397 : H.264/AAC movie deadlocks with totem in gstreamer code, p...
- * 339935 : [adder] dead-locks when adding sink pads in PAUSED state
-
-Changes since 0.10.6:
-
- * typefind improvements
- * bug-fixes in textoverlay, audioconvert, videotestsrc,
- multifdsink and audio source/sink base classes
- * Ice-cast metadata support has moved from gnomevfssrc to the
- icydemux element in gst-plugins-good
- * audioresample now supports floating point samples
- * Adder element fixes.
- * Fixes for network playback and audio resampling in playbin
-
-Bugs fixed since 0.10.6:
-
- * 340060 : [adder] handle newsegment events properly
- * 340375 : [API 0.11] [patch] typefind to differentiate between mp4 ...
- * 339405 : [textoverlay] can't display '\n' character
- * 338657 : [patch] adder should send events from src-pad to all sink...
- * 338919 : [patch] alsasink should also query witdh capabilities fro...
- * 301759 : [audioresample] float audio support (for OSX audio sinks)
- * 331901 : [videotestsrc] framerate=0/1 gives assertion error
- * 333657 : Replacing icy demuxing in gnomevfssrc
- * 336339 : [audioresample] should support width != 16
- * 338718 : [patch] [audioconvert] correctly clip float samples > 1.0
- * 338778 : [patch] Bad audio with ASX files
- * 338991 : [patch] Videoscale doesn't pass on pixel-aspect ratio
- * 339574 : [patch] Race condition in multifdsink can lead to spuriou...
- * 339786 : [typefinding] wavpack typefinding doesn't always work
- * 340369 : [volume element] " volume " property range insufficient
- * 340379 : [playbin] doesn't insert audioresample, causes problems w...
- * 340392 : Problem with internal-decodebin
- * 341160 : [multifdsink] client_status enum has an uninitialized nick
- * 341182 : Accessing playbin's streaminfo property from high languag...
- * 341432 : [playbin] automatically get icecast metadata requiring ic...
- * 341542 : some users have an assertion failed: (GST_VIDEO_SINK_WIDT...
- * 341557 : Map GST_TAG_IMAGE < = > ID3v2 APIC tag
-
-API added since 0.10.6:
-
- * client-fd-removed signal added to multifdsink
- * stream-info-value-array property added to playbin
- * gst_video_calculate_display_ratio() in libgstvideo
-
-Changes since 0.10.5:
-
- * QoS in sinks and transform elements
- * Needs GStreamer 0.10.5 for new GstBaseSink::async_playback() vmethod
- * added theoraparse element
-
-Bugs fixed since 0.10.5:
-
- * 313136 : [playbin] hang while playing truncated ogg file
- * 172848 : [subparse] subtitles with special chars are displayed as ...
- * 305279 : [riff] uncompressed AVIs with 24bpp don't work
- * 320765 : [ffmpegcolorspace] make win32+msvc compliant, don't use _...
- * 323852 : Disable tests/icles on platforms that do not have X
- * 325653 : build errors compiling audioresample on win32(vs7)
- * 327357 : gst-plugins-base fails to compile with GCC 4.1
- * 334620 : [gnomevfssrc] fails to connect to icecast streaming servers
- * 334822 : [ffmpegcolorspace] YVU9 support
- * 335028 : [typefinding] ID3 v1 tag is not recognized with mp3-in-wa...
- * 335365 : inefficient use of GList in gst-plugins-base
- * 336190 : [gnomevfssink] should accept non-URI filenames as " location "
- * 336194 : [gnomevfssrc] some minor memory leaks
- * 336477 : plugins need better/univied descriptions
- * 336617 : Unable to recognise MPEG TS stream
- * 337548 : Memory leaks in basertpdepayload
- * 337945 : [oggdemux] segment stop position ignored
- * 338419 : Regression in the handling of files with multiple audio/s...
- * 338897 : Videoscale crashes as part of DVD to Ogg transcoding
- * 339013 : [videorate] Goes into an infinite loop
- * 339047 : [riff] handle H264 fourcc in addition to h264
- * 339212 : ISO file typefinding regression
- * 330748 : deadlock in base audio sink on playing- > paused state change
-
-Bugs fixed since 0.10.4:
-
- * 334216 : [gnomevfssrc] won't open some media on NFS mounts any longer
- * 334226 : typefindfunctions plugin crashes on PPC on registration
-
-Changes since 0.10.3:
-
- * (Experimental) QoS support
- * oggmuxer now creates 100% valid streams for Theora, Vorbis and Speex
- * documentation updates
- * better support for subtitles (seeking)
-
-Bugs fixed since 0.10.3:
-
- * 310202 : [subtitles] < i > < /i > tags and others should be supported i...
- * 312439 : XVideo output doesn't work on remote displays (probably r...
- * 321271 : audio output is truncated at EOS
- * 321650 : Can't decode this ogm file
- * 325732 : [oggdemux] problem when seeking to time less than 4s with...
- * 325972 : [typefinding] doesn't recognise this mp3
- * 326720 : [alsasink] doesn't support more than 2 channels anymore
- * 330711 : [ffmpegcolorspace] problems with palettized RGB (fencount...
- * 330789 : gstbaseaudiosink causes noise on seeking
- * 330888 : Fix build with gcc 2.95 (again)
- * 331295 : gnomevfssink doesn't respect umask when creating files
- * 331526 : 3GP type detection is too simple
- * 331678 : Decodebin is not reusable within a single pipeline (as in...
- * 331690 : playbin won't play my last.fm stream
- * 331763 : [alsamixer] unmute sets the volume to 100%
- * 331765 : [alsamixer] mixer applet slider doesn't want to move from...
- * 331903 : [videorate] doesnt handle input caps of framerate=0/1 sanely
- * 332778 : [ogmparse] " Already an existing pad " WARNING
- * 332964 : random crashes in mp3_type_find
- * 333254 : theora encoder does not set IN_CAPS flag properly
- * 333352 : [gnomevfssink] reports disk full as generic error
- * 333488 : Allow for palette < 256 colours in AVI files
- * 333510 : [PATCH] Fix gst_pad_new_from_template (gst_static_pad_tem...
- * 333545 : [riff] set depth on wma caps to make asfdemux and pitfdll...
- * 333663 : [patch] unref the result of gst_pad_get_parent
- * 333900 : [typefind] cannot play a particular mp3 file
- * 334112 : variable not initialized
- * 334129 : Disable frame dropping for now
- * 317038 : use default channel layout if none is specified in multic...
- * 319340 : [cdparanoia] uncorrected-error signal never fired
-
-API added since 0.10.3:
-
- * GstTextOverlay::halignment
- * GstTextOverlay::valignment
-
-Changes since 0.10.2:
-
- * typefind improvements
- * Ogg decoding and encoding fixes
- * Improved audio and video sink classes
- * Bug and leak fixes
- * Improved video scaling
- * On-the-fly visualisation switching
- * Subtitle support
-
-Bugs fixed since 0.10.2:
-
- * 330244 : gsttextoverlay.c:895: 'struct _GstCollectData' has no mem...
- * 324000 : [playbin] post error or message on unknown input
- * 153004 : [typefind] can't identify mp3 file with one single mpeg f...
- * 323874 : [playbin] leaks sinks and threads when using gconfaudiosink
- * 324626 : ffmpegcolorspace support for fourcc " UYVY "
- * 326447 : check that all elements in -base pass queries they can't ...
- * 328263 : Fix build with gcc 2.95
- * 328279 : [decodebin] timeout issue when pre-rolling
- * 329326 : Fix oggmux removing pads from collect pads
-
-Changes since 0.10.1:
-
- * ported gnomevfssink, cdparanoia
- * New library and base class: GstCddaBaseSrc
- * ported mixerutils.h
- * added 'sine-tab' waveform to audiotestsrc
- * added float audio to audiorate
-
-Bugs fixed since 0.10.1:
-
- * 324216 : [cdparanoia] missing patches from 0.8
- * 324696 : [videotestsrc] does not start counting the time from zero...
- * 324900 : Problem compiling gst-plugins-base with Forte
- * 325984 : [playbin] cannot handle sources that produce raw audio/video
- * 325990 : patch videotestsrc for using glib types
- * 326601 : GstRingBuffer crashes with alaw/mulaw caps
- * 327114 : [theoradec] should post tags on the bus
- * 327216 : vorbisdec segfaults on certain queries
-
-API added since 0.10.1:
-
- * added libgstcddabase
- * added mixerutils.h
-
-Changes since 0.10.0:
-
- * Parallel installability with 0.8.x series
- * Threadsafe design and API
- * removed gst-launch-ext
- * Ported: ogmparse
- * Fixes for: subparse, xvimagesink, audioresample, videorate, decodebin
-
-Bugs fixed since 0.10.0:
-
- * 322347 : GstBaseRtpDepayload timestamps are wring
- * 323900 : Basertpdepayloader lets NEWSEGMENT events through unfiltered
- * 323878 : missing < string.h > inclusion (for memset & FD_ZERO)
-
-API added since 0.10.0:
-
- * GstAlsaMixer::device
- * GstAlsaMixer::device-name
-
-Bugs fixed since 0.9.7:
-
- * 319172 : gstreamer-plugins-base-0.9.pc doesn't export linking flags
- * 323017 : While(1) loop with sleep(0) in basertpdepayload.c
-
-Changes since 0.9.6:
-
- * Parallel installability with 0.8.x series
- * Threadsafe design and API
- * ximagesink and xvimagesink updates and interactive test
- * added pango
- * rename net to netbuffer library
- * rtp element renaming
- * stream selector fixes
-
-Bugs fixed since 0.9.6:
-
- * 319618 : [decodebin] some ogg videos don't play
- * 320644 : RTP packetizer does't set the packet timestamps correctly
- * 322388 : xvimagesink force-aspect-ratio=True always displays squar...
- * 322704 : oggdemux typefind list leak
-
-Changes since 0.9.5:
-
- * Parallel installability with 0.8.x series
- * Threadsafe design and API
- * lots of leak fixes
- * flicker-free and rewritten X sinks
- * fractional framerates
- * removed sinesrc, replaced by audiotestsrc
-
-Bugs fixed since 0.9.5:
-
- * 316442 : playbin should use autoaudiosink/autovideosink by default
- * 318353 : [ffmpegcolorspace] forward-port fixes from 0.8 branch
- * 320200 : vorbisenc: min-bitrate and max-bitrate are 1/1000 bps rat...
- * 321164 : gstringbuffer stops working under load
- * 321426 : ximage plugin should be renamed to ximagesink
- * 321446 : sinesrc should be dropped in favour of audiotestsrc
- * 321451 : GstRtpBuffer: no way to create a sub buffer with only the...
- * 321816 : [API] xoverlay API to post prepare-xwindow-id message
- * 321894 : vorbisenc doesn't compile
- * 322117 : Rename libgsttagedit to libgsttag
-
-Changes since 0.9.4:
-
- * video caps now use a good range for framerate and w/h
- * oggdemux/oggmux improvements
- * playbin improvements
-
-Bugs fixed since 0.9.4:
-
- * 319110 : [PATCH] oggdemux chain finding is slow
- * 320058 : playbin of a jpeg over http does not work
- * 320923 : [volume] doesn't build on Solaris
- * 321011 : gstbasertpdepayload doesn't send the " new segment " event ...
-
-Changes since 0.9.3:
-
- * New element: audiotestsrc
- * typefind improvements
- * buffer-frames removed
-
-Changes since 0.9.2:
-
- * RTP base classes
-
-Bugs fixed since 0.9.2:
-
- * 313251 : ximagesink unused functions
- * 315159 : audioconvert lost 24 bit conversions in the rewrite
+This is GStreamer Base Plugins 1.2.0
+
+Changes since 1.0:
+
+New API:
+ • GstContext negotiation / sharing / announcing for sharing a
+ generic context between elements, e.g. a display handle
+ • GL texture upload conversion meta for allowing different
+ buffer types to be converted to an OpenGL texture
+ • GstCapsFeatures as extension to GstCaps for allowing the
+ negotiation of specific memory or meta requirements between
+ elements
+ • GstMemory flags for contiguous and non-mappable memory
+ • The stream-start event has optional flags now, e.g. for signalling
+ sparse streams
+ • The stream-start even has an optional group-id field now to signal
+ all streams that should be played together
+ • Allocators library in gst-plugins-base, currently only with generic
+ dmabuf memory support
+ • insertbin library for easier handling of dynamically linked
+ pipelines (in -bad for now)
+ • EGL helper library (in -bad for now)
+ • MPEG-TS data structure library (in -bad for now)
+ • New GstVideoRegionOfInterestMeta to describe a region of interest on
+ video frames.
+ • GstVideoDecoder/Encoder has new ::flush() vfunc to replace the
+ ill-defined ::reset() vfunc.
+ • The URI query allows to query the redirected URI now.
+
+Major changes:
+ • New tool: gst-play-1.0 in gst-plugins-base for basic playback
+ testing on the command line.
+ • New plugins:
+ ∘ mssdemux for Microsoft Smooth Streaming
+ ∘ dashdemux for DASH adaptive streaming protocol
+ ∘ bluez for interaction with Bluetooth devices
+ ∘ openjpeg for JPEG2000 decoding and encoding
+ ∘ daala for experimental Daala decoding and encoding
+ ∘ vpx plugin has experimental VP9 decoding and encoding support
+ ∘ webp plugin for WebP decoding (encoding to be added later)
+ ∘ Various others: yadif, srtp, sbc, fluidsynth, midiparse,
+ mfc, ivtv, accuraterip and audiofxbad
+
+ • Moved plugins:
+ ∘ dtmf, vp8rtp, scaletempo and rtpmux plugins are in
+ gst-plugins-good now
+
+ • Video:
+ ∘ Fix handling of interlaced video in converters such as videoscale
+ and videoconvert (e.g. scale both fields independently)
+ ∘ videoconvert will try harder to minimise quality losses when
+ conversion is necessary
+ ∘ The experimental GstSurfaceConverter, GstSurfaceMeta and
+ GstVideoContext APIs from the (confusingly-named)
+ libgstbasevideo-1.0 library in gst-plugins-bad have now been
+ removed and been replaced by new APIs in GStreamer Core and
+ gst-plugins-base (see above). Since that was all that was left in
+ this library, the entire experimental libgstbasevideo-1.0 library
+ has been removed from gst-plugins-bad
+ ∘ Chroma subsampling and chroma siting conversion is better handled
+ in videoconvert and the support for interlaced video was improved.
+ ∘ New pinwheel and spoke patterns in videotestsrc
+ ∘ videomixer can now accept different video formats on its sinkpads
+ and converts to a common format during mixing
+
+ • Audio:
+ ∘ audioconvert will try harder to minimise quality losses when
+ conversion is necessary
+ ∘ adder now allows muting/unmuting of its input streams, and also
+ per-input stream volume
+ ∘ pulseaudio elements can switch between devices during playback now
+ ∘ aacparse can convert between ADTS←→RAW
+
+ • Platform specific changes:
+ ∘ Caps, events, etc. are now printed in the GStreamer debug logs
+ with their content instead of just the pointer address even on
+ non-glibc platforms (e.g. Windows, OSX, Android).
+ ∘ Network elements (UDP/TCP) now work better with platforms,
+ where IPv6 sockets can't handle IPv4 (e.g. Windows)
+ ∘ Linux/BSD: v4l2 had many improvements and cleanups
+
+ • Other changes:
+ ∘ gst-libav now uses libav 9
+ ∘ Static linking of plugins is supported now (also in 1.0.7)
+ ∘ rtspsrc: add support for NetClientClock: when the server suggests a
+ GstNetTimeProvider in the SDP, set up a GstNetClientClock that
+ slaves to the remote clock and suggest this clock in provide_clock.
+ Simplifies synchronized playback of a resource from an RTSP server.
+ gst-rtsp-server now supports adding this to the SDP and can provide
+ a network clock
+ ∘ RTP retransmission / NACK support and big RTP jitterbuffer improvements
+ ∘ SRTP and DTLS support
+ ∘ Changes to many elements and core to use the correct sticky event
+ order and also not lose any important sticky events during flushing
+ ∘ >1000 fixed bug reports, and many other bug fixes and other
+ improvements everywhere that had no bug report
+
+Things to look out for:
+ • Single header includes for all libraries, e.g. #include
+ <gst/video/video.h> - this was needed for some bindings.
+ • Stricter (correct) caps subset checking in some cases where this was
+ not correct before. Caps will now always fail to be a compatible
+ subset of another set of caps if the subset caps are missing some
+ fields that the superset caps have. This might lead to not-negotiated
+ errors if caps are incomplete now. However, it also prevents possible
+ data corruption caused by piping data formatted in an
+ incompatible/unexpected way into some elements. Check your h264 caps
+ for stream-format and alignment fields and AAC caps for the
+ stream-format field. This change will also be included in the next
+ stable 1.0.8 release.
+ • Stricter checking for missing events and correct sticky event order
+ (stream-start, caps, segment) in some places; this is not enabled in
+ stable releases by default, but you may get warnings when using git
+ builds, development releases or when compiling with
+ -UG_DISABLE_ASSERT in CFLAGS
+ • x264enc now outputs data in byte-stream by default if downstream has
+ ANY caps (e.g. appsink without caps set, filesink, udpsink,
+ tcpserversink etc.)
+ • The MPEG TS demuxer posts messages contain the PMT, PAT, etc. in a
+ different format now. This new format uses the data structures from
+ the new MPEGTS library
+ • The GstContext API has changed between 1.1.4 and 1.1.90
projects / platform / upstream / gstreamer.git / blobdiff