mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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1005 lines
48 KiB
Text
1005 lines
48 KiB
Text
This is GStreamer Base Plug-ins 0.10.22, "Hidey Hidey Hidey Ho"
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IMPORTANT NOTES
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1) Please note that decodebin2 and playbin2 API included in this release is
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still considered unstable and WILL change in future releases. At this stage,
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only developers or early adopters should consider using decodebin2 or playbin2
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API embodied in their signals and properties.
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Changes since 0.10.21:
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* Require gettext 0.17
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* Replace audioresample with speexresample from -bad
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* Support new formats in RIFF: uncompressed RGB, WMA lossless, VP6
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* Move libgstapp and elements from -bad
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* Support color-key setting and probing for Xv properties
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* Improve typefinding for various formats
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* Extend audio sinks for pull-mode operation
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* Support for more subtitle formats
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* More development on decode2bin and playbin2
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* RTP and SDP fixes
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* Many bug fixes and improvements
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Bugs fixed since 0.10.21:
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* 562163 : theoraenc likely ignoring segments
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* 562258 : rtspsrc element takes long time to error out if the addre...
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* 561789 : [volume] deadlocks with a controller attached
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* 554533 : [xvimagesink] allow setting colorkey if possible
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* 567511 : colorkey in xvimagesink gets reset when element is reused
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* 116051 : libresample doesn't handle > factor of 2 rate conversion
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* 346218 : [audioresample] doesn't do anti aliasing
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* 385061 : [audioresample?] investigate high CPU usage
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* 456788 : [subparse] can't handle UTF-16 charset encoded subtitle.
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* 525807 : [vorbisenc] vorbisenc has problems with a gnlsource that ...
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* 546955 : gstoggmux EOS handling issue
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* 549417 : [audioresample] unit test fails on 64bit linux
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* 549510 : audioresample doesn't negotiate ideal caps
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* 552237 : UTF-16 srt confuses gstreamer, misdetected as mp3
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* 552559 : Implementation of SLAVE_SKEW in baseaudiosrc
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* 552569 : audioresample producing strange sized buffers
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* 552801 : audioconvert can overflow with big audio buffers
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* 554879 : Add ability to specify format for date/time display in Gs...
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* 555257 : Doesn't display srt subtitles saved with BOM
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* 555319 : add FFV1 fourcc to riff-media
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* 555607 : subrip subtitles typefind too strict
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* 555699 : [PATCH] theoradec: prefer container's pixel aspect ratio ...
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* 556025 : build failure in tests/icles
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* 556066 : Last byte of FLAC image buffer chopped off
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* 557365 : subparse check fails
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* 558124 : [PLUGIN-MOVE] Move speexresample as audioresample2 to -base
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* 559111 : ALSA sink hangs on USB audio device unplug while playing
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* 559478 : does not play windows media streams correctly
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* 559567 : `gst_base_audio_sink_sync_latency' should call `gst_base_...
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* 561436 : videorate element add image/jpeg to caps template
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* 561734 : playbin2 additions
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* 561780 : Playbin2 should work without volume too
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* 561924 : oggdemux hangs when given corrupt input via non-seekable ...
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* 562270 : build without gdk fails
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* 563143 : ximagesink/xvimagesink : _alloc_buffer returns non-clean ...
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* 563174 : Implement gst_rtcp_packet_remove
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* 563508 : [rgvolume] Unit test fails with passthrough assertions
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* 563718 : Theora check out of date
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* 563904 : GNOME Goal: Clean up GLib and GTK+ includes
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API added since 0.10.21:
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* clockoverlay::time-format
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* GstRingBuffer:gst_ring_buffer_activate()
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* GstRingBuffer:gst_ring_buffer_is_active()
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* GstRingBuffer:gst_ring_buffer_convert()
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* Add GST_TYPE_BASE_AUDIO_(SRC|SINK)_SLAVE_METHOD to the public API
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* gst_netaddress_get_address_bytes()
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* gst_netaddress_set_address_bytes()
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Changes since 0.10.20:
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* Continue playbin2 development
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* Ogg improvements - CELT support, skeleton fixes
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* DVD subpicture support
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* Improved audio dithering random number generator
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* xvimagesink/ximagesink fixes
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* Vorbis encoding and decoding fixes
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* Recognise Kate subtitle streams
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* Many bug-fixes and enhancements
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Bugs fixed since 0.10.20:
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* 537380 : [gnomevfssrc] Doesn't handle short reads properly
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* 538656 : xvimagesink support for autofill/colorkey property
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* 540334 : Build fails without X in tests/examples/seek
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* 528299 : Multiple GstMixerTracks with the same label cause problem...
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* 530068 : Ogg Streams with Skeleton and Granulepos > 0 do not work(...
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* 537009 : playbin2 silly typo breaks signals
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* 537045 : decodebin2 sometimes emits 'drained' multiple times
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* 537599 : [oggdemux] skeleton streams not skipped in ogg
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* 537889 : [xvimagesink] colorbalance is bad
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* 538232 : vorbisenc/vorbisdec don't work with a live source
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* 538663 : gdppay memleak in gst_gdp_pay_reset
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* 540215 : decodebin does not insert a queue for raw data type
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* 540351 : [avidemux] Doesn't know about Duck DK4 ADPCM
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* 540497 : ffmpegcolorspace is returning wrong size
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* 541358 : cross mingw32 gcc: getaddrinfo is not in ws2_32.dll befor...
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* 544306 : rtspsrc debug=1 segfaults with some libc
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* 548898 : GStreamer-CRITICAL errors on seeking beyond stream borders
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* 548913 : vorbisenc being picky about rounding errors in timestamps
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* 549062 : Video devices aren't updated on subsequent probing.
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* 549814 : [typefind] add application/pdf typefinder
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* 550582 : [oggdemux] KATE streams not recognised
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* 550638 : [typefind] Recognize some jpeg2k file types
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* 550656 : recognize TrueSpeech in wavparse
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* 550729 : gst-plugins-base won't compile with " -pedantic " option
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* 552960 : tagdemux asserts and aborts on truncated files
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* 553244 : theoraparse doesn't work at all (throws criticals and ass...
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API added since 0.10.20:
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* Add "index" property to GstMixerTrack to differantiate between
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multiple mixer tracks with the same label.
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Changes since 0.10.19:
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* RTP improvements
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* Support digest auth for RTSP
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* Additional documentation
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* Support DSCP QoS in multifdsink
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* Add NV12/NV21 video buffer layouts
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* Video scaling now bilinear by default
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* Support more than 8 channels in audio conversions
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* Channel mapping fixes for audioconvert
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* Improve tmplayer and sami subtitle support
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* Support 1x1 pixel buffers for videoscale
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* Typefinding improvements for MPEG2, musepack
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* Ogg/Dirac mapping updated in oggmux
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* Fixes in ogg demuxing
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* audiosink synchronisation and slaving fixes
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* Support muting of the audio in playbin by selecting -1 as the audio stream
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* Work done on playbin2 and uridecodebin
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* Improvements in the experimental GIO plugin
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* decodebin fixes
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* Handle GAP buffers in some places
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* Various other leak and bug-fixes
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Bugs fixed since 0.10.20:
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* 526794 : [giosrc] totem doesn't work with some gvfs backends
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* 510417 : [PLUGIN-MOVE] Move gio to gst-plugins-base
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* 509125 : crash in CD Player: - playing CD - lowering/...
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* 517813 : [audioconvert] make gap aware
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* 302798 : [playbin] add mute property
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* 342294 : Setting playbin property current-audio=-1 also stops the ...
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* 398033 : [audioconvert] support more than 8 channels
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* 419351 : [avi/a52dec] AV synchronization problems
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* 467911 : [subparse] sami parser update
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* 469933 : multifdsink IPv6 and diffserv TOS/TC markup
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* 506659 : [textoverlay] rendering error when using non-standard widths
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* 512333 : [gstvorbistag] Retrieve Ogg/Vorbis cover art as image met...
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* 512382 : [playbin] race condition when pausing/playing multiple in...
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* 518037 : pbutils-enumtypes.c is not included in win32/vs6/libgstpb...
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* 521761 : gstaudioclock frozen the clock value until reaches latest...
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* 522401 : gdpdepay doesn't validate payload CRCs
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* 523993 : playbin2 blocks after a while when listening to a radio s...
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* 524724 : [PATCH] [baseaudiosrc] buffer-time and latency-time do no...
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* 525665 : Crash on Ogg/Vorbis with chain=NULL
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* 525915 : [streamheader] Unit test fails with " gst_adapter_peek: as...
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* 526173 : [typefinding] fails to detect mpeg video stream whereas m...
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* 529018 : gst_ogm_parse_stream_header creates fraction value with w...
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* 529500 : [videotestsrc] support for NV12 and NV21
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* 529546 : [Playbin] Memory leak in streaminfo handling
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* 530068 : Ogg Streams with Skeleton and Granulepos > 0 do not work(...
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* 530531 : [typefinding] bad read in mpeg_video_stream_type_find
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* 530719 : gst_video_calculate_display_ratio fails when playing Ogg ...
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* 530962 : [subparse] parses only every second line of TMPlayer subt...
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* 532454 : [NV12/NV21] videotestsrc and ffmpegcolorspace don't play ...
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* 533087 : GstRTSPTransport kept opaque in docs
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* 533817 : [audioconvert] Can't use default 7 channel layout / only ...
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* 534071 : Gdppay memleak
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* 534331 : race in decodebin when changing states while the internal...
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* 535356 : vorbisdec doesn't support 8 channels
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* 536475 : gdppay memleak and possible crash
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* 536521 : Refcounting errors in playbin
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* 536874 : Build failure on windows
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* 532166 : [ffmpegcolorspace] support NV12 format
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* 533617 : [audioconvert] Produces silence when converting 1/2 chann...
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* 536848 : [giosrc] Doesn't handle short reads properly
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* 536849 : [giosrc] Very slow doing any playback
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* 518082 : [alsamixer] playback volumes overwritten by capture volum...
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* 435633 : [PATCH] videorate not (fully) segment aware; causes frame...
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* 532364 : tcpclientsrc broken in 0.10.19
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* 533075 : gst_rtp_buffer_compare_seqnum doesn't do what it says
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* 533265 : [cddabasesrc] Sound Juicer cut a sector when ripping a track
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API additions since 0.10.20:
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* decodebin2::sink-caps property
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* giosrc::file property
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* giosink::file property
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* gst_base_audio_src_set_slave_method()
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* gst_base_audio_src_get_slave_method()
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* GstAudioClock::gst_audio_clock_reset()
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* GstBaseAudioSrc:actual-buffer-time property
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* GstBaseAudioSrc:actual-latency-time property
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* gst_audio_check_channel_positions()
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* add gst_tag_image_data_to_image_buffer()
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* add gst_tag_list_add_id3_image()
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* add GST_TAG_IMAGE_TYPE_NONE enum value
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Changes since 0.10.18:
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* Handle EAGAIN when polling sockets in rtspconnection
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Changes since 0.10.17:
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* Experimental GIO plugin
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* Continued playbin2 development
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* RTP fixes
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* Better network element support on Windows
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* Various other bug-fixes and improvements
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Bugs fixed since 0.10.17:
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* 509637 : [API] [basertpaudiopayload] add _set_samplebits_options()
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* 510229 : [gnomevfssrc] HTTPS support
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* 511478 : [rtpbuffer] add gst_rtp_buffer_set_extension_data function
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* 511810 : [RTSP] Uses MT-unsafe gmtime() function
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* 512899 : [alsa] gstalsasink.c:527: warning: 'snd_pcm_sw_params_set...
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* 513167 : Fix compiler warning due to disabled signals in mixertrac...
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* 514307 : [playbin] warning in nautilus, volume element can't be cr...
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* 514623 : Ogg Theora video slow
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* 514937 : Correct initialization of hints in is_multicast_address()
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* 515654 : xvimagesink doesn't build with --disable-xshm
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* 516246 : [alsasink] handle negative delay from snd_pcm_delay
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* 517420 : typefind: add h264 elementary stream discovery
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* 517991 : problems with configure file depending on GCC compiler
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* 518039 : libgstrtsp MSVC 6.0 compile error
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* 518162 : [subparse] handle italic text starting with " / " with Micr...
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* 518940 : [playbin2] make _get_*_tags() match vfuncs prototype in c...
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* 519906 : [API] add GstMixerOptions::get_values vfunc
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* 519916 : [API] add mixer-changed and options-list-changed messages
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* 520523 : [API] Unreviewed changes to ringbuffer API
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* 521743 : libgstnetbuffer.def exports not up to date
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* 522625 : [video] gst_video_format_parse_caps() broken for RGBA for...
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* 523054 : gstbasesrc crashes when called from typefind helpers
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* 511825 : [RTSP] compiler warning on FreeBSD
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* 520300 : [alsasrc] provide-clock=false messes up buffer durations
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API added since 0.10.17:
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* GstRTPBuffer:gst_rtp_buffer_set_extension_data()
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* add GST_VIDEO_FORMAT_Y41B and GST_VIDEO_FORMAT_Y42B.
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* add GstMixerOptions::get_values vfunc (#519906)
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* add gst_mixer_options_list_changed(), gst_mixer_mixer_changed() and
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gst_mixer_message_parse_options_list_changed(). Fixes #519916.
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* gst_base_rtp_audio_payload_set_samplebits_options()
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* GstNetBuffer::gst_netaddress_equal
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Changes since 0.10.16:
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* Work-around ABI breakage due to unfortunate use of the
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GST_DISABLE_DEPRECATED macro
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* Export 2 missing functions needed for bindings in the win32 build
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* Initialise the GstRingBuffer GType from a thread-safe context
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Bugs fixed since 0.10.16:
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* 511825 : [RTSP] compiler warning on FreeBSD
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* 513018 : crash in Volume Control: I typed my password at t...
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* 512334 : g_critical() when using GstAudioFilter & GST_DEBUG
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Changes since 0.10.15:
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* Handle newer Theora granule-pos semantics
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* Introducing first alpha version playbin2 - the upcoming successor to
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playbin
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* Fixes in playbin handling of stream-switching
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* New API for uniform handling of raw-video format buffers.
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* Improvements for RTSP/RTP handling
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* RIFF lib additions for VC-1 and AVC1 fourccs
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* Many other bug-fixes and improvements
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Bugs fixed since 0.10.15:
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* 506132 : Review of changes in video/video.h
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* 320984 : [oggdemux] cannot handle multiple chains
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* 373011 : [playbin] throws error when switching off subtitles
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* 436756 : Intermittent crashes in Pidgin in audioclock g_type_class...
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* 462740 : [streamselector] patch to improve default stream selection
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* 486840 : [alsamixer] use _all variants when setting the mixer
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* 497964 : theoraenc test fails
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* 498228 : gst-plugins-base-0.10.15 does not compile on FreeBSD (Gen...
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* 499697 : Provide better pkg-config files
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* 502497 : [subparse] SubRip subtitles starting from 0 not recognised
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* 503440 : The control sockets used by gstrtspconnection.c are never...
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* 503930 : [cdda] warning: 'eos' may be used uninitialized in this f...
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* 506928 : [alsamixer] add " PCM " as master fall back for cards that ...
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* 508138 : [decodebin] does not error out if pad activation fails
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* 509762 : missing file in win32/MANIFEST
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* 511274 : gst_rtp_buffer_get_extension_data is returning FALSE when...
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* 496731 : [PATCH] xvimagesink leaks memory if initialization fails
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* 496761 : [PATCH] RTSP message leaks memory when uninitialized
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* 500763 : SIGSEGV while playing ogg audio file
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API additions since 0.10.15:
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* New GstVideoFormat API and helper functions in libgstvideo
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* gst_base_audio_sink_set_provide_clock()
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* gst_base_audio_sink_get_provide_clock()
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* gst_base_audio_sink_set_slave_method()
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* gst_base_audio_sink_get_slave_method()
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* gst_base_audio_src_set_provide_clock()
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* gst_base_audio_src_get_provide_clock()
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Changes since 0.10.14:
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* RTP/RTSP/RTCP/SDP support improved
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* New FFT support library libgstfft, based on Kiss FFT
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* New formats supported in volume and audiotestsrc
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* Fixes in audiorate and videorate
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* Audio capture fixes
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* Playbin and decodebin fixes
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* New tagdemux base class for ID3/APE style tag readers
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* Fix a nasty crash in the X sinks on shutdown
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* New tags supported
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* Add support for multichannel WAV files.
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* Preserve channel layout information when up/down-mixing.
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* Many bug-fixes and improvements
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Bugs fixed since 0.10.14:
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* 475395 : decodebin2 leaks request-pads
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* 475451 : [decodebin2] leaks ghostpad
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* 378770 : [xvimagesink] race condition in event thread?
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* 407282 : [decodebin2] autoplug-sort signal has GList ** parameter
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* 430677 : [audioconvert] does not preserve channel positions when f...
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* 442654 : [volume] controller bypassed by default
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* 445529 : [volume] support for 24/32-bit audio/x-raw-int
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* 446766 : return code for gst_base_rtp_payload_audio_handle_event()
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* 451970 : Subparse requires HTML parser
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* 453650 : [audiobasesrc] two alsasrcs do not work in one pipeline
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* 459334 : [textoverlay] expose pango line alignment property
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* 459585 : [basertpdepayload] api without namespace
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* 460422 : [audiotestsrc] Add support for float and double output
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* 462805 : [alsa] compilation fails with gcc 4.2
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* 462979 : Add 'silent' property to GstTimeOverlay
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* 463215 : [audioconvert] compile errors
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* 464320 : [PATCH] gst-plugins-base-0.14 does not build for win32
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* 464666 : [playbin] QT trailer hangs in preroll with decodebin2
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* 464690 : Add connection-speed property to uridecodebin element
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* 465015 : [playbin] Not removed probes causes deadlocks in streamin...
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* 465028 : some warnings with mingw
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* 467667 : GST_FRAMES_TO_CLOCK_TIME() and GST_CLOCK_TIME_TO_FRAMES()...
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* 468129 : [basertpaudiopayload] event handler returns the wrong value
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* 468619 : New library gstfft: FFT library for integer and float typ...
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* 470456 : [API] add gst_missing_*_installer_detail_new()
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* 470766 : [ssaparse] line breaks in SSA subtitle parser
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* 471067 : Make the SDP code useable for generating SDP descriptions
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* 471194 : [rtpbuffer] RTP headers are wrong for win32
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* 473097 : [baseaudiosink] gstreamer-properties hangs when testing s...
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* 474384 : gstrtsp-enumtypes.c and .h needed for win32
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* 474880 : [xvimagesink] [ximagesink] leaking buffer caps reference
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* 475731 : rtspconnection is able to read incomplete messages
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* 483620 : All Rtp buffers are discarded -- gst_rtp_buffer_get_payl...
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* 484989 : memleak, not unrefed caps for gstbasertppayload.c
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* 489010 : Please change default channel order for WAVE_EXT-less .wa...
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* 491722 : [playbin] regression: crash with external subtitles
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* 492098 : [GstFFT] Broken scaling
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* 492114 : Build issues on Windows/MSVC
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* 492306 : compilation errors with MinGW
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* 492813 : Missing symbols in libgstrtp.def
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* 493986 : Build issues on Windows (missing symbols)
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* 494346 : pre-release vs6 patch
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* 496548 : Including malloc.h breaks macos build
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* 496724 : DSW file references non-existent DSP files
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* 464079 : audiotestsrc doesn't respond to conversion queries properly
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* 442065 : floatcast.h includes config.h and might break other apps
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* 466717 : gst_event_new_new_segment_full:assertion `start < = stop' ...
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* 485753 : Decodebin2 deadlocks when nulling pipeline during typefind
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* 464028 : Move connection-speed from playbin to playbasebin
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API added since 0.10.14:
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* GstTagDemux base class for simple tag demuxers
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* GstBaseAudioSrc::provide-clock property
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* gst_rtcp_ntp_to_unix()
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* gst_rtcp_unix_to_ntp()
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* 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
|
|
|