gstreamer/NEWS
Jan Schmidt f2c0a73133 Release 0.10.17
Original commit message from CVS:
Release 0.10.17
2008-01-30 14:19:05 +00:00

780 lines
37 KiB
Text

This is GStreamer Base Plug-ins 0.10.17, "Peanut Butter and Jelly"
IMPORTANT NOTES
This release is identical to 0.10.16, with several small but significant bug
fixes. The most important one fixes crashes in gnome-volume-control and other
consumers of the GstMixer API, due to an unfortunate ABI break. Everyone
should use 0.10.17 instead of 0.10.16 in all cases.
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.
2) On some systems, the current release of gst-plugins-good (0.10.6) may fail to
build against this release of gst-plugins-base with an error like:
gstid3v2mux.cc:547: error: 'GST_TAG_MUSICBRAINZ_SORTNAME' was not declared in this scope
In this case, you should either patch the configure file of gst-plugins-good to
remove -DGST_DISABLE_DEPRECATED from DEPRECATED_CFLAGS=, or else compile
with make DEPRECATED_CFLAGS=''
3) Some users may experience problems using the 'mp3parse' element from the
previous gst-plugins-ugly release (0.10.6). This is due to a bug in mp3parse
exposed by changes in decodebin in gst-plugins-base. It will be fixed in the
upcoming release of gst-plugins-ugly next month. In the meantime as a
workaround, you can set the rank of mp3parse to GST_RANK_NONE in
gst-plugins-ugly/gst/mpegaudioparse/gstmpegaudioparse.c when compiling, or
or remove the /usr/lib/gstreamer-0.10/libgstmpegaudioparse.so file entirely.
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