This is GStreamer Base Plug-ins 0.10.25, "Standard disclaimers apply"

Changes since 0.10.24:

      * Add per-stream volume controls

      * Theora 1.0 and Y444 and Y42B format support

      * Improve audio capture timing

      * GObject introspection support

      * Improve audio output startup

      * RTSP improvements

      * Use pango-cairo instead of pangoft2

      * Allow cdda://(device#)?track URI scheme in cddabasesrc

      * Support interlaced content in videoscale and ffmpegcolorspacee

      * Many other bug fixes and improvements

Bugs fixed since 0.10.24:

      * 595401 : gobject assertion and null access to volume instance in playbin

      * 563828 : [decodebin2] Complains about loops in the graph when demuxer output requires another demuxer

      * 591677 : Easy codec installation is not working

      * 588523 : smarter sink selection in playbin2

      * 590146 : adder regressions

      * 321532 : [cddabasesrc] Support device setting in cdda:// URI

      * 340887 : add pangocairo textoverlay plugin.

      * 397419 : [oggdemux] ogm video with subtitles stuck on first frame

      * 556537 : [PATCH] typefind: more flexible MPEG4 start code recognition

      * 559049 : gstcheck.c:76:F:general:test_state_changes_* failure: GST_IS_CLOCK(clock) assertion fails

      * 567660 : [API] need a stream volume interface for sinks that do volume control

      * 567928 : Make videorate work with a live source

      * 571610 : [playbin] Scale of volume property is not documented

      * 583255 : [playbin2] deadlock when disabling visualisations

      * 586180 : RTSP improvements

      * 588717 : [oggmux] gst_caps_unref() warning if not linked downstream

      * 588761 : [videoscale] Needs special support for interlaced content

      * 588915 : audioresample's output offset counter's initialization could maybe be improved

      * 589095 : [appsrc] clarify documentation on caps and linkage

      * 589574 : [typefind] incorrect sdp file detection

      * 590243 : [videoscale] Claims to support MAX width/height

      * 590425 : Slaved alsasrc clock with slave-method=re-timestamp not usable for RTP audio

      * 590856 : [decodebin2] triggers assertion failure on NULL caps

      * 591207 : totem does display the following subtitle srt file.

      * 591357 : gst-plugins-base git won't build due to warning in gstrtspconnection.c

      * 591577 : [playbin2] Incorrect error message string

      * 591664 : [playbin2] after seeking, srt subtitles don't resync correctly

      * 591934 : timestamp drift in audioresample

      * 592544 : Remove regex.h check

      * 592657 : [appsink] Blocks after entering on pause state

      * 592864 : deadlocks from recent inputselector/streamselector change

      * 592884 : [playbin2] g_object_get increases refcount by 2 and therefore leaves memleak

      * 593035 : gdp doesn't preserve fields of the buffers put into the caps' streamheader

      * 593284 : basertppayloader takes time in instance init

      * 594020 : Totem don't play videos from ssh remote host

      * 594094 : Playback Error playing Midi file

      * 594136 : [alsasink] Regression from 0.10.23 -- element reuse doesn't work

      * 594165 : [theoraenc] Implement support for new formats

      * 594256 : improved slave-skew resynch mechanism

      * 594258 : missing break in rtcpbuffer

      * 594275 : Add cast to navigation to fix compiler warning

      * 594623 : Expose playsink as a fully-fledged element

      * 594732 : parse error

      * 594757 : build fails due to warning in gstbasertppayload.c

      * 594993 : [introspection] pkg-config file madness

      * 594994 : [streamvolume] Add get_type function to the documentation

      * 595454 : [cddabasesrc] uri format change breaks rhythmbox

      * 545807 : [baseaudiosink] audible crack when starting the pipeline

API added since 0.10.24:

      * gst_rtsp_connection_create_from_fd()

      * gst_rtsp_connection_set_http_mode()

      * gst_rtsp_watch_write_data()

      * gst_rtsp_watch_send_message()

      * GstBaseRTPPayload::perfect-rtptime

      * GstBaseRTPAudioPayload::gst_base_rtp_audio_payload_flush()

      * GstVideoSinkClass::show_frame()

      * GstVideoSink:show-preroll-frame

      * GST_MIXER_TRACK_READONLY

      * GST_MIXER_TRACK_WRITEONLY

      * GstStreamVolume interface

Changes since 0.10.23:

      * Recognise Kate subpicture subtitles

      * Support progressive download in playbin2

      * GIO improvements

      * Add buffer-list support in appsink

      * Add gaussian-noise mode to audiotestsrc

      * bump cdparanoia req to 0.10.2 and improve caching

      * Improve audio source base class

      * Add frame-by-frame stepping and examples

      * Extend stream-probing in decodebin2

      * Many RTSP improvements

      * support for PGS subpictures

      * adder improvements

      * Add Y444, v210, v216 formats

      * implement preset interface in vorbisenc, theoraenc, oggmux

      * Improve libvisual visualisation timestamp tracking

      * playbin2 enhancements: custom audiosink, subpictures, cdda

      * Improvements in textrender

      * Support raw YUV 4:2:2 and SIREN in RIFF

      * Add 4:2:2 and 4:4:4 support to theoradec

      * Many other bug-fixes and improvements

Bugs fixed since 0.10.23:

      * 510417 : [gio] make non-experimental

      * 513373 : [PATCH] [gstvorbistag] Preserve cover art in Ogg/Vorbis tags

      * 529300 : [giosink] [PATCH] Allow overwrite

      * 531035 : [cdparanoia] Should depend on LGPL'd version of the libra...

      * 567997 : [patch] add allow-pull-scheduling property to audio sinks

      * 576552 : [subparse] post GST_TAG_SUBTITLE_CODEC tags

      * 577637 : [playbin2] expose temp-location property

      * 579692 : mp3_type_find is over-optimistic

      * 580318 : [tagdemux] drops tag events from upstream

      * 581460 : [baseaudiosrc] Reusing audio source leads to null timesta...

      * 581571 : ARGB and alignment added to textrender

      * 582021 : autogen: libtoolize must be called before aclocal

      * 582749 : uridecodebin caps property not implemented yet

      * 582819 : multifdsink: add num-fds property

      * 583867 : gdpdepay + identity cause failed assertions

      * 584020 : [playbin2] inadvertently resets configured audio/video sinks

      * 584686 : [playbin2] Need {audio,video,text}-tags-changed signals

      * 585197 : [subparse] fails to detect subrip subtitles with fewer th...

      * 585758 : Remove deprecated GTK+ symbols

      * 585970 : gst_audioringbuffer_get_type is not thread safe

      * 585994 : gst-rtsp-message doesn't support " Timestamp " filed

      * 586331 : [cdparanoia] expose cd cache size parameter

      * 586356 : [playbin2] use private copy of input-selector as long as ...

      * 586519 : white Gaussian noise would be useful in audiotestsrc

      * 587080 : rtsp fails to compile - doesn't see some ws2tcpip functions

      * 587278 : Support for GstBufferList in appsink

      * 587676 : Call tzset() before localtime_r(), in e.g. gst-plugins-ba...

      * 587695 : Patches to add stream-status messages audio elements

      * 587896 :  " No stream given yet " error from giostreamsrc

      * 587980 : gstchannelmix.c: protect debug code with GST_DISABLE_GST_...

      * 588078 : [playbin2] Fails to go to READY again after an error

      * 588205 : Pipeline with giostreamsrc will not enter playing state

      * 588550 : build failure in git, missing gstinterfaces-0.10

      * 588551 : queue2: download buffering fixes

      * 588724 : [vorbisdec] empty encoder string causes GStreamer

      * 588746 : [audiotestsrc] Make sure tags are properly serialized in ...

      * 588747 : [adder] Serialize incoming in-band events (tags) in the d...

      * 588748 : [adder] Check dataflow consistency in unit tests

      * 589075 : [playbin2] changing volume doesn't work after stream rest...

      * 589581 : typefinder: recognise more Kate subtitle categories

      * 589622 : Cannot use both playbin and input-selector

      * 589663 : gstreamer asserts in gstaudiofilter

      * 589797 : alsasrc does not set GstAlsaSrc- > handle to NULL after snd...

      * 590470 : [typefinding] certain flac-in-ogg files not detected any ...

      * 536313 : [cdda] Remove sha1 copy once we depend on glib-2.16

      * 579642 : [oggdemux] handle broken ogg/vorbis files better

      * 582528 : playbin2 Audio CD playback broken since

      * 583318 : Assertion from within playbin2

      * 585079 : undefined references to gst_adapter_* functions in schro

      * 585708 : [adder] Wrong handling of flushing seeks

      * 588218 : Siren in .wav support

      * 586920 : rtsp: needs < netinet/in.h > on FreeBSD

API added since 0.10.23:

      * GstNetAddress::gst_netaddress_to_string()

      * Add gst_rtsp_watch_queue_data()

      * playbin2: Add {audio,video,text}-tags-changed signals

      * Add gst_color_balance_get_balance_type()

      * Add gst_mixer_get_mixer_type()

Changes since 0.10.22:

      * New navigation API to support DVD playback

      * playbin2 improvements

      * RTSP extensions to allow extra headers and options

      * Replace audioresampler with speexresample based code

      * Support interlacing flags in the gstvideo library

      * Support new RIFF formats

      * Improve typefinding

      * Support more frame formats in videoscale

      * Many other bug-fixes and improvements

Bugs fixed since 0.10.22:

      * 577637 : [playbin2] expose temp-location property

      * 580120 : [playbin2] unit test fails

      * 478512 : [alsamixer] volume control slider not working

      * 574962 : rhythmbox crash in flac_type_find

      * 564139 : Documentation of TCP plugins

      * 577436 : xvimagesink should use xcontext- > depth and not count bits...

      * 350311 : [playbin2] support for subpicture subtitles

      * 378094 : Enable pango elements to handle UYVY

      * 543591 : Gnonlin can not play theora streams

      * 553295 : [riff] fuzzed AVI file causes segfault

      * 565105 : Gstreamer does not change from READY back to PAUSED in sa...

      * 565777 : [riff] unrecognised video fourcc 0x10000002 for mpeg2 in avi

      * 566661 : [typefind] Fall back to file extension using uri query

      * 567255 : [riff] doesn't detect codec_id 0x706d as AAC (amongst other)

      * 567636 : [pbutils] Missing plugins code shouldn't ask for the same...

      * 567740 : bogus warning in decodebin2?

      * 568482 : linking problems in gst-plugins-base

      * 569655 : [ffmpegcolorspace] Add UYVY422 to GRAY8 conversion function

      * 570142 : Documentation is broken for uridecodebin

      * 570356 : aac typefinder failure

      * 570768 : [ximagesink] wrong mouse pointer position if output windo...

      * 570832 : Add flags to enhance mixer interfaces

      * 571009 : [tagdemux] WMA file with id3v2 tag causes assertion to fail

      * 571147 : [ffmpegcolorspace/videotestsrc] Add support for packed/pl...

      * 572577 : [playbin2] deadlock on shutdown

      * 572872 : [ffmpegcolorspace] Add YVYU colorspace

      * 572993 : [subparse] broken libregex dependency on Windows

      * 573165 : Generate additional export files for gstreamer app plugin

      * 573528 : Wrong format modifier in gstgiobasesink.c

      * 573529 : In gstrtspconnection.c some functions are called with wro...

      * 574293 : [decodebin2] deadlock on shutdown

      * 574319 : Missing HAVE_PROCESS_H in win32/common/config.h

      * 574447 : gstadder.c: line 904: error C2036: 'gpointer' : unknown size

      * 574939 : [typefinding] flac typefinder mis-typefinds PDFs as flac ...

      * 575550 : srt subtitle file keeps playbin2 from playing

      * 575638 : kissfft copyright

      * 575649 : [oggdemux] duration query in time format returns true wit...

      * 576019 : On Windows queue2 can't write files longer than 2-4 GiB, ...

      * 576142 : [vorbisenc] Non-header output buffers have NULL caps

      * 576180 : [playbin2] Uses unref'd audiosink volume if using gconfau...

      * 576586 : [alsamixer] gnome-sound-properties freeze

      * 577054 : [videoscale] Not valgrind clean

      * 577709 : Review new navigation API

      * 577827 : [appsink] Have appsink new_buffer-callback return GstFlow...

      * 578583 : [PATCH] multifdsink doesn't handle sync-method=latest-key...

      * 578656 : Implement upstream GstForceKeyUnit events in theoraenc

      * 579129 : pkgconfig: appsrc/appsink can not be linked to uninstalled

      * 579130 : app: expose trivial type macros

      * 579192 : gst_rtcp_packet_get_type should not assert on packet content

      * 579203 : baseaudiosink: unparenting the ringbuffer in NULL causes ...

      * 579267 : [rtspconnection] g_async_queue_new_full() is GLib-2.16 AP...

      * 579463 : [cddabasesrc] [cdparanoiasrc] no longer emits discid

      * 579668 : audioresample fails to build with --disable-gst-debug

      * 579734 : [playbin] raw_decoding_mode seems to be set unconditionally

      * 579912 : [decodebin2] multiqueue is too small in time (interleave ...

      * 580470 : [audioresample] causes pipelines to go out of sync and be...

      * 580952 : [audioresample] bad quality/pops compared to plughw

      * 581727 : [playbin2] make playsink go to PAUSED async

      * 569682 : playbin2 leaks request pad from input selector

      * 580020 : [vorbisenc] causes buffers to be out of segment if new se...

      * 562794 : rtspsrc fails to create a socket on Win32 sometimes.

      * 567396 : playbin2: DECODE_BIN_LOCK occasionally called twice withi...

      * 567982 :  " queued_bytes " field isn't updated while flushing the que...

      * 571299 : [appsink] Handoff callback API

      * 574443 : rtsp win32 - forgotten variable

      * 574516 : [typefind] add typefinder for photoshop .psd files

      * 574964 : gst_app_src_end_of_stream(), mutex on error return

      * 575256 : rtspsrc fails to resolve hostnames

      * 575588 : decodebin2 deadlock

      * 576187 : [playbin2] Stalls video sink when disabling subtitles in ...

      * 576188 : [playbin2] Reusing a playbin2 instance with visualization...

      * 576190 : [playbin2] Deadlock when reusing playbin2 after an error

      * 577288 :  " Internal playbin error " when seeking to the end of files

      * 577610 : RTCP feedback messages support in GstRTCPPacket

      * 577794 : [playbin2] leaks elements set through properties

      * 578118 : [multifdsink] add option to not resend the streamheader w...

      * 578506 : Pipeline with alsasrc and alsasink cannot change state ba...

      * 578942 : Missing RTSP headers related to Windows Media extension.

      * 580271 : videorate: fails to clear discont flag on duplicated buffers

      * 580649 : uridecodebin: bug on documentation published in website

API added since 0.10.22:

      * GstRTSP::gst_rtsp_options_as_text()

      * GstRTSPMessage::gst_rtsp_message_take_header()

      * GstRTSPRange::gst_rtsp_range_to_string()

      * New Navigation interface commands, queries and messages

      * gst_rtsp_channel_new()

      * gst_rtsp_channel_unref()

      * gst_rtsp_channel_attach()

      * gst_rtsp_channel_queue_message()

      * gst_rtsp_connection_accept()

      * GstAppSink::gst_app_sink_set_callbacks()

      * GST_VIDEO_FORMAT_YVYU,GST_VIDEO_BUFFER_TFF,GST_VIDEO_BUFFER_RFF,GST_VIDEO_BUFFER_ONEFIELD

      * GST_MIXER_FLAG_HAS_WHITELIST,GST_MIXER_FLAG_GROUPING,GST_MIXER_TRACK_NO_RECORD,GST_MIXER_TRACK_NO_MUTE,GST_MIXER_TRACK_WHITELIST

      * GstAppSrc::emit-signals

      * GstAppSrc::gst_app_src_set_emit_signals()

      * GstAppSrc::gst_app_src_get_emit_signals()

      * GstAppSrc::gst_app_src_set_callbacks()

      * RTSP::gst_rtsp_connection_get_url()

      * GstRTSPLowerTrans::GST_RTSP_LOWER_TRANS_HTTP

      * RTSP:gst_rtsp_connection_set_tunneled()

      * RTSP:gst_rtsp_connection_is_tunneled()

      * RTSP::gst_rtsp_connection_set_ip()

      * RTSP::gst_rtsp_connection_get_tunnelid()

      * RTSP::gst_rtsp_connection_do_tunnel()

      * RTSP::gst_rtsp_watch_reset()

IMPORTANT NOTES

1) Please note that decodebin2 and playbin2 API included in this release is

still considered unstable and WILL change in future releases. At this stage,

only developers or early adopters should consider using decodebin2 or playbin2

API embodied in their signals and properties.

Changes since 0.10.21:

      * Require gettext 0.17

      * Replace audioresample with speexresample from -bad

      * Support new formats in RIFF: uncompressed RGB, WMA lossless, VP6

      * Move libgstapp and elements from -bad

      * Support color-key setting and probing for Xv properties

      * Improve typefinding for various formats

      * Extend audio sinks for pull-mode operation

      * Support for more subtitle formats

      * More development on decode2bin and playbin2

      * RTP and SDP fixes

      * Many bug fixes and improvements

Bugs fixed since 0.10.21:

      * 562163 : theoraenc likely ignoring segments

      * 562258 : rtspsrc element takes long time to error out if the addre...

      * 561789 : [volume] deadlocks with a controller attached

      * 554533 : [xvimagesink] allow setting colorkey if possible

      * 567511 : colorkey in xvimagesink gets reset when element is reused

      * 116051 : libresample doesn't handle > factor of 2 rate conversion

      * 346218 : [audioresample] doesn't do anti aliasing

      * 385061 : [audioresample?] investigate high CPU usage

      * 456788 : [subparse] can't handle UTF-16 charset encoded subtitle.

      * 525807 : [vorbisenc] vorbisenc has problems with a gnlsource that ...

      * 546955 : gstoggmux EOS handling issue

      * 549417 : [audioresample] unit test fails on 64bit linux

      * 549510 : audioresample doesn't negotiate ideal caps

      * 552237 : UTF-16 srt confuses gstreamer, misdetected as mp3

      * 552559 : Implementation of SLAVE_SKEW in baseaudiosrc

      * 552569 : audioresample producing strange sized buffers

      * 552801 : audioconvert can overflow with big audio buffers

      * 554879 : Add ability to specify format for date/time display in Gs...

      * 555257 : Doesn't display srt subtitles saved with BOM

      * 555319 : add FFV1 fourcc to riff-media

      * 555607 : subrip subtitles typefind too strict

      * 555699 : [PATCH] theoradec: prefer container's pixel aspect ratio ...

      * 556025 : build failure in tests/icles

      * 556066 : Last byte of FLAC image buffer chopped off

      * 557365 : subparse check fails

      * 558124 : [PLUGIN-MOVE] Move speexresample as audioresample2 to -base

      * 559111 : ALSA sink hangs on USB audio device unplug while playing

      * 559478 : does not play windows media streams correctly

      * 559567 : `gst_base_audio_sink_sync_latency' should call `gst_base_...

      * 561436 : videorate element add image/jpeg to caps template

      * 561734 : playbin2 additions

      * 561780 : Playbin2 should work without volume too

      * 561924 : oggdemux hangs when given corrupt input via non-seekable ...

      * 562270 : build without gdk fails

      * 563143 : ximagesink/xvimagesink : _alloc_buffer returns non-clean ...

      * 563174 : Implement gst_rtcp_packet_remove

      * 563508 : [rgvolume] Unit test fails with passthrough assertions

      * 563718 : Theora check out of date

      * 563904 : GNOME Goal: Clean up GLib and GTK+ includes

API added since 0.10.21:

      * clockoverlay::time-format

      * GstRingBuffer:gst_ring_buffer_activate()

      * GstRingBuffer:gst_ring_buffer_is_active()

      * GstRingBuffer:gst_ring_buffer_convert()

      * Add GST_TYPE_BASE_AUDIO_(SRC|SINK)_SLAVE_METHOD to the public API

      * gst_netaddress_get_address_bytes()

      * gst_netaddress_set_address_bytes()

Changes since 0.10.20:

      * Continue playbin2 development

      * Ogg improvements - CELT support, skeleton fixes

      * DVD subpicture support

      * Improved audio dithering random number generator

      * xvimagesink/ximagesink fixes

      * Vorbis encoding and decoding fixes

      * Recognise Kate subtitle streams

      * Many bug-fixes and enhancements

Bugs fixed since 0.10.20:

      * 537380 : [gnomevfssrc] Doesn't handle short reads properly

      * 538656 : xvimagesink support for autofill/colorkey property

      * 540334 : Build fails without X in tests/examples/seek

      * 528299 : Multiple GstMixerTracks with the same label cause problem...

      * 530068 : Ogg Streams with Skeleton and Granulepos > 0 do not work(...

      * 537009 : playbin2 silly typo breaks signals

      * 537045 : decodebin2 sometimes emits 'drained' multiple times

      * 537599 : [oggdemux] skeleton streams not skipped in ogg

      * 537889 : [xvimagesink] colorbalance is bad

      * 538232 : vorbisenc/vorbisdec don't work with a live source

      * 538663 : gdppay memleak in gst_gdp_pay_reset

      * 540215 : decodebin does not insert a queue for raw data type

      * 540351 : [avidemux] Doesn't know about Duck DK4 ADPCM

      * 540497 : ffmpegcolorspace is returning wrong size

      * 541358 : cross mingw32 gcc: getaddrinfo is not in ws2_32.dll befor...

      * 544306 : rtspsrc debug=1 segfaults with some libc

      * 548898 : GStreamer-CRITICAL errors on seeking beyond stream borders

      * 548913 : vorbisenc being picky about rounding errors in timestamps

      * 549062 : Video devices aren't updated on subsequent probing.

      * 549814 : [typefind] add application/pdf typefinder

      * 550582 : [oggdemux] KATE streams not recognised

      * 550638 : [typefind] Recognize some jpeg2k file types

      * 550656 : recognize TrueSpeech in wavparse

      * 550729 : gst-plugins-base won't compile with " -pedantic " option

      * 552960 : tagdemux asserts and aborts on truncated files

      * 553244 : theoraparse doesn't work at all (throws criticals and ass...

API added since 0.10.20:

      * Add "index" property to GstMixerTrack to differantiate between

        multiple mixer tracks with the same label.

Changes since 0.10.19:

      * RTP improvements

      * Support digest auth for RTSP

      * Additional documentation

      * Support DSCP QoS in multifdsink

      * Add NV12/NV21 video buffer layouts

      * Video scaling now bilinear by default

      * Support more than 8 channels in audio conversions

      * Channel mapping fixes for audioconvert

      * Improve tmplayer and sami subtitle support

      * Support 1x1 pixel buffers for videoscale

      * Typefinding improvements for MPEG2, musepack

      * Ogg/Dirac mapping updated in oggmux

      * Fixes in ogg demuxing

      * audiosink synchronisation and slaving fixes

      * Support muting of the audio in playbin by selecting -1 as the audio stream

      * Work done on playbin2 and uridecodebin

      * Improvements in the experimental GIO plugin

      * decodebin fixes

      * Handle GAP buffers in some places

      * Various other leak and bug-fixes

Bugs fixed since 0.10.20:

      * 526794 : [giosrc] totem doesn't work with some gvfs backends

      * 510417 : [PLUGIN-MOVE] Move gio to gst-plugins-base

      * 509125 : crash in CD Player: - playing CD - lowering/...

      * 517813 : [audioconvert] make gap aware

      * 302798 : [playbin] add mute property

      * 342294 : Setting playbin property current-audio=-1 also stops the ...

      * 398033 : [audioconvert] support more than 8 channels

      * 419351 : [avi/a52dec] AV synchronization problems

      * 467911 : [subparse] sami parser update

      * 469933 : multifdsink IPv6 and diffserv TOS/TC markup

      * 506659 : [textoverlay] rendering error when using non-standard widths

      * 512333 : [gstvorbistag] Retrieve Ogg/Vorbis cover art as image met...

      * 512382 : [playbin] race condition when pausing/playing multiple in...

      * 518037 : pbutils-enumtypes.c is not included in win32/vs6/libgstpb...

      * 521761 : gstaudioclock frozen the clock value until reaches latest...

      * 522401 : gdpdepay doesn't validate payload CRCs

      * 523993 : playbin2 blocks after a while when listening to a radio s...

      * 524724 : [PATCH] [baseaudiosrc] buffer-time and latency-time do no...

      * 525665 : Crash on Ogg/Vorbis with chain=NULL

      * 525915 : [streamheader] Unit test fails with " gst_adapter_peek: as...

      * 526173 : [typefinding] fails to detect mpeg video stream whereas m...

      * 529018 : gst_ogm_parse_stream_header creates fraction value with w...

      * 529500 : [videotestsrc] support for NV12 and NV21

      * 529546 : [Playbin] Memory leak in streaminfo handling

      * 530068 : Ogg Streams with Skeleton and Granulepos > 0 do not work(...

      * 530531 : [typefinding] bad read in mpeg_video_stream_type_find

      * 530719 : gst_video_calculate_display_ratio fails when playing Ogg ...

      * 530962 : [subparse] parses only every second line of TMPlayer subt...

      * 532454 : [NV12/NV21] videotestsrc and ffmpegcolorspace don't play ...

      * 533087 : GstRTSPTransport kept opaque in docs

      * 533817 : [audioconvert] Can't use default 7 channel layout / only ...

      * 534071 : Gdppay memleak

      * 534331 : race in decodebin when changing states while the internal...

      * 535356 : vorbisdec doesn't support 8 channels

      * 536475 : gdppay memleak and possible crash

      * 536521 : Refcounting errors in playbin

      * 536874 : Build failure on windows

      * 532166 : [ffmpegcolorspace] support NV12 format

      * 533617 : [audioconvert] Produces silence when converting 1/2 chann...

      * 536848 : [giosrc] Doesn't handle short reads properly

      * 536849 : [giosrc] Very slow doing any playback

      * 518082 : [alsamixer] playback volumes overwritten by capture volum...

      * 435633 : [PATCH] videorate not (fully) segment aware; causes frame...

      * 532364 : tcpclientsrc broken in 0.10.19

      * 533075 : gst_rtp_buffer_compare_seqnum doesn't do what it says

      * 533265 : [cddabasesrc] Sound Juicer cut a sector when ripping a track

API additions since 0.10.20:

      * decodebin2::sink-caps property

      * giosrc::file property

      * giosink::file property

      * gst_base_audio_src_set_slave_method()

      * gst_base_audio_src_get_slave_method()

      * GstAudioClock::gst_audio_clock_reset()

      * GstBaseAudioSrc:actual-buffer-time property

      * GstBaseAudioSrc:actual-latency-time property

      * gst_audio_check_channel_positions()

      * add gst_tag_image_data_to_image_buffer()

      * add gst_tag_list_add_id3_image()

      * add GST_TAG_IMAGE_TYPE_NONE enum value

Changes since 0.10.18:

      * Handle EAGAIN when polling sockets in rtspconnection

Changes since 0.10.17:

      * Experimental GIO plugin

      * Continued playbin2 development

      * RTP fixes

      * Better network element support on Windows

      * Various other bug-fixes and improvements

Bugs fixed since 0.10.17:

      * 509637 : [API] [basertpaudiopayload] add _set_samplebits_options()

      * 510229 : [gnomevfssrc] HTTPS support

      * 511478 : [rtpbuffer] add gst_rtp_buffer_set_extension_data function

      * 511810 : [RTSP] Uses MT-unsafe gmtime() function

      * 512899 : [alsa] gstalsasink.c:527: warning: 'snd_pcm_sw_params_set...

      * 513167 : Fix compiler warning due to disabled signals in mixertrac...

      * 514307 : [playbin] warning in nautilus, volume element can't be cr...

      * 514623 : Ogg Theora video slow

      * 514937 : Correct initialization of hints in is_multicast_address()

      * 515654 : xvimagesink doesn't build with --disable-xshm

      * 516246 : [alsasink] handle negative delay from snd_pcm_delay

      * 517420 : typefind: add h264 elementary stream discovery

      * 517991 : problems with configure file depending on GCC compiler

      * 518039 : libgstrtsp MSVC 6.0 compile error

      * 518162 : [subparse] handle italic text starting with " / " with Micr...

      * 518940 : [playbin2] make _get_*_tags() match vfuncs prototype in c...

      * 519906 : [API] add GstMixerOptions::get_values vfunc

      * 519916 : [API] add mixer-changed and options-list-changed messages

      * 520523 : [API] Unreviewed changes to ringbuffer API

      * 521743 : libgstnetbuffer.def exports not up to date

      * 522625 : [video] gst_video_format_parse_caps() broken for RGBA for...

      * 523054 : gstbasesrc crashes when called from typefind helpers

      * 511825 : [RTSP] compiler warning on FreeBSD

      * 520300 : [alsasrc] provide-clock=false messes up buffer durations

API added since 0.10.17:

      * GstRTPBuffer:gst_rtp_buffer_set_extension_data()

      * add GST_VIDEO_FORMAT_Y41B and GST_VIDEO_FORMAT_Y42B.

      * add GstMixerOptions::get_values vfunc (#519906)

      * add gst_mixer_options_list_changed(), gst_mixer_mixer_changed() and

        gst_mixer_message_parse_options_list_changed(). Fixes #519916.

      * gst_base_rtp_audio_payload_set_samplebits_options()

      * GstNetBuffer::gst_netaddress_equal

Changes since 0.10.16:

      * Work-around ABI breakage due to unfortunate use of the

        GST_DISABLE_DEPRECATED macro

      * Export 2 missing functions needed for bindings in the win32 build

      * Initialise the GstRingBuffer GType from a thread-safe context

Bugs fixed since 0.10.16:

      * 511825 : [RTSP] compiler warning on FreeBSD

      * 513018 : crash in Volume Control: I typed my password at t...

      * 512334 : g_critical() when using GstAudioFilter & GST_DEBUG

Changes since 0.10.15:

      * Handle newer Theora granule-pos semantics

      * Introducing first alpha version playbin2 - the upcoming successor to

	playbin

      * Fixes in playbin handling of stream-switching

      * New API for uniform handling of raw-video format buffers.

      * Improvements for RTSP/RTP handling

      * RIFF lib additions for VC-1 and AVC1 fourccs

      * Many other bug-fixes and improvements

Bugs fixed since 0.10.15:

      * 506132 : Review of changes in video/video.h

      * 320984 : [oggdemux] cannot handle multiple chains

      * 373011 : [playbin] throws error when switching off subtitles

      * 436756 : Intermittent crashes in Pidgin in audioclock g_type_class...

      * 462740 : [streamselector] patch to improve default stream selection

      * 486840 : [alsamixer] use _all variants when setting the mixer

      * 497964 : theoraenc test fails

      * 498228 : gst-plugins-base-0.10.15 does not compile on FreeBSD (Gen...

      * 499697 : Provide better pkg-config files 

      * 502497 : [subparse] SubRip subtitles starting from 0 not recognised

      * 503440 : The control sockets used by gstrtspconnection.c are never...

      * 503930 : [cdda] warning: 'eos' may be used uninitialized in this f...

      * 506928 : [alsamixer] add " PCM " as master fall back for cards that ...

      * 508138 : [decodebin] does not error out if pad activation fails

      * 509762 : missing file in win32/MANIFEST

      * 511274 : gst_rtp_buffer_get_extension_data is returning FALSE when...

      * 496731 : [PATCH] xvimagesink leaks memory if initialization fails

      * 496761 : [PATCH] RTSP message leaks memory when uninitialized

      * 500763 : SIGSEGV while playing ogg audio file

API additions since 0.10.15:

      * New GstVideoFormat API and helper functions in libgstvideo

      * gst_base_audio_sink_set_provide_clock()

      * gst_base_audio_sink_get_provide_clock()

      * gst_base_audio_sink_set_slave_method()

      * gst_base_audio_sink_get_slave_method()

      * gst_base_audio_src_set_provide_clock()

      * gst_base_audio_src_get_provide_clock()

Changes since 0.10.14:

      * RTP/RTSP/RTCP/SDP support improved

      * New FFT support library libgstfft, based on Kiss FFT

      * New formats supported in volume and audiotestsrc

      * Fixes in audiorate and videorate

      * Audio capture fixes

      * Playbin and decodebin fixes

      * New tagdemux base class for ID3/APE style tag readers

      * Fix a nasty crash in the X sinks on shutdown

      * New tags supported

      * Add support for multichannel WAV files.

      * Preserve channel layout information when up/down-mixing.

      * Many bug-fixes and improvements

Bugs fixed since 0.10.14:

      * 475395 : decodebin2 leaks request-pads

      * 475451 : [decodebin2] leaks ghostpad

      * 378770 : [xvimagesink] race condition in event thread?

      * 407282 : [decodebin2] autoplug-sort signal has GList ** parameter

      * 430677 : [audioconvert] does not preserve channel positions when f...

      * 442654 : [volume] controller bypassed by default

      * 445529 : [volume] support for 24/32-bit audio/x-raw-int

      * 446766 : return code for gst_base_rtp_payload_audio_handle_event()

      * 451970 : Subparse requires HTML parser

      * 453650 : [audiobasesrc] two alsasrcs do not work in one pipeline

      * 459334 : [textoverlay] expose pango line alignment property

      * 459585 : [basertpdepayload] api without namespace

      * 460422 : [audiotestsrc] Add support for float and double output

      * 462805 : [alsa] compilation fails with gcc 4.2

      * 462979 : Add 'silent' property to GstTimeOverlay

      * 463215 : [audioconvert] compile errors

      * 464320 : [PATCH] gst-plugins-base-0.14 does not build for win32

      * 464666 : [playbin] QT trailer hangs in preroll with decodebin2

      * 464690 : Add connection-speed property to uridecodebin element

      * 465015 : [playbin] Not removed probes causes deadlocks in streamin...

      * 465028 : some warnings with mingw

      * 467667 : GST_FRAMES_TO_CLOCK_TIME() and GST_CLOCK_TIME_TO_FRAMES()...

      * 468129 : [basertpaudiopayload] event handler returns the wrong value

      * 468619 : New library gstfft: FFT library for integer and float typ...

      * 470456 : [API] add gst_missing_*_installer_detail_new()

      * 470766 : [ssaparse] line breaks in SSA subtitle parser

      * 471067 : Make the SDP code useable for generating SDP descriptions

      * 471194 : [rtpbuffer] RTP headers are wrong for win32

      * 473097 : [baseaudiosink] gstreamer-properties hangs when testing s...

      * 474384 : gstrtsp-enumtypes.c and .h needed for win32

      * 474880 : [xvimagesink] [ximagesink] leaking buffer caps reference

      * 475731 : rtspconnection is able to read incomplete messages

      * 483620 : All Rtp buffers are discarded --  gst_rtp_buffer_get_payl...

      * 484989 : memleak, not unrefed caps for gstbasertppayload.c

      * 489010 : Please change default channel order for WAVE_EXT-less .wa...

      * 491722 : [playbin] regression: crash with external subtitles

      * 492098 : [GstFFT] Broken scaling

      * 492114 : Build issues on Windows/MSVC

      * 492306 : compilation errors with MinGW

      * 492813 : Missing symbols in libgstrtp.def

      * 493986 : Build issues on Windows (missing symbols)

      * 494346 : pre-release vs6 patch

      * 496548 : Including malloc.h breaks macos build

      * 496724 : DSW file references non-existent DSP files

      * 464079 : audiotestsrc doesn't respond to conversion queries properly

      * 442065 : floatcast.h includes config.h and might break other apps

      * 466717 : gst_event_new_new_segment_full:assertion `start < = stop' ...

      * 485753 : Decodebin2 deadlocks when nulling pipeline during typefind

      * 464028 : Move connection-speed from playbin to playbasebin

API added since 0.10.14:

      * GstTagDemux base class for simple tag demuxers

      * GstBaseAudioSrc::provide-clock property

      * gst_rtcp_ntp_to_unix()

      * gst_rtcp_unix_to_ntp()

      * gst_rtp_buffer_get_header_len()

      * gst_rtp_buffer_get_extension_data()

      * gst_rtp_buffer_compare_seqnum()

      * gst_rtp_buffer_ext_timestamp()

      * gst_rtcp_packet_sdes_copy_entry()

      * gst_install_plugins_supported()

      * gst_missing_*_installer_detail_new() convenience API

      * gst_rtsp_connection_poll()

      * GstTextOverlay::line-alignment property

Changes since 0.10.13:

      * Audio dither and noise-shaping when reducing bit-depth

      * RTSP and SDP helper libraries added

      * Experimental buffering element "queue2" now supports pull-mode

        and file-based buffering.

      * Support for more 32-bit video pixel layouts

      * Various fixes and improvements

Bugs fixed since 0.10.13:

      * 380625 : [x*imagesink] add 'handle-expose' property

      * 385527 : oggmux sometimes gets DELTA flag on output wrong near start

      * 402076 : videoscale 4-tap method broken for downscaling

      * 437169 : [xvimagesink] add property to disable Xv double-buffering

      * 441264 : queue2 support to do buffering on a file

      * 442553 : [v4lsrc] doesn't output segments in GST_FORMAT_TIME

      * 442557 : [videorate] doesn't handle latency queries

      * 442944 : Audiotestsrc can overflow on seeks

      * 444523 : [queue2] Pull mode support

      * 444630 : Compilation error with fsseko (from gstqueue2.c) -- unabl...

      * 445505 : [queue2] It does not work in pull mode with oggdemux

      * 446551 : [queue2] Buffering is not working properly if it is set t...

      * 446572 : [queue2] Division by zero

      * 446972 : warning when compiling  gstoggdemux.c

      * 449156 : Regression in CVS for decodebin2

      * 450875 : Missing files in po/POTFILES.in

      * 451707 : [tag] UTF-8 in ID3v1 tag not correctly decoded

      * 451908 : [ffmpegcolorspace] regression: doesn't accept GST_VIDEO_C...

      * 454264 : Playbin fails to " play " image url after a movie url

      * 456656 : [API] Addition of audio buffer clipping function to gstaudio

      * 460978 : gst_audio_buffer_clip outputs warnings

      * 152864 : [PATCH] GstAlsaMixer doesn't support signals

      * 360246 : [audioconvert] Optionally apply dithering

      * 394061 : Add support for Subviewer subtitles

      * 420326 : Base payloader class has wrong property types and ranges

      * 451145 : [vorbisdec] errors out on 0-sized packets

      * 459204 : [PATCH] [playbin] gst_play_base_bin_get_streaminfo_value_...

API added since 0.10.13:

      * RTSP and SDP libraries added

      * gst_rtsp_base64_decode_ip

      * Add buffer clipping function gst_audio_buffer_clip for raw audio 

        buffers. Fixes #456656.

      * gst_mixer_get_mixer_flags

      * gst_mixer_message_parse_mute_toggled

      * gst_mixer_message_parse_record_toggled

      * gst_mixer_message_parse_volume_changed

      * gst_mixer_message_parse_option_changed

      * GstMixerMessageType

      * GstMixerFlags

Changes since 0.10.12:

      * Many fixes and improvements

      * RTP and RTCP support improved

Bugs fixed since 0.10.12:

      * 339838 : [audioconvert] support floats with non-native endianness

      * 393975 : closing x/xvimagesink window crashes gst-launch

      * 405072 : [API] add gst_tag_freeform_string_to_utf8()

      * 413799 : [subparse] add support for MPL2 format

      * 414645 : GstMixerTrack should make untranslated label available

      * 420079 : [audioconvert] Uses biased rounding which results in dist...

      * 420578 : [subparse] add more colour map in sami parser

      * 421834 : videorate breaks on dimension changes

      * 423051 : Vorbis tags of type double use locale-dependent formatting

      * 423055 : Verify ReplayGain vorbistag processing in libs/tag testsuite

      * 425455 : Decodebin2 leaks pads

      * 426250 : GstPlayBaseBin leaks streaminfo objects

      * 428187 : Rtp base depayloader class doesn't send new_segment after...

      * 431672 : gst_base_rtp_audio_payload_push() should take object of i...

      * 432362 : [ximagesink] doesn't build if XShm is not available

      * 432755 : [videorate] leaks buffer if flow != OK

      * 432984 : [baseaudiosrc] misleading warning message when dropping s...

      * 433888 : [theoradec] does not generate a perfect stream

      * 436562 : Theoradec doesn't work well with gnonlin

      * 438840 : [theoradec] does not compile with old version of libtheora

      * 440997 : [gstriff] Doesn't handle width!=depth files with audio/x-...

      * 441295 : audioconvert doesn't build on VS6

      * 442024 : regression in playbin buffering

      * 350299 : [playbin] " Internal data flow error " opening movie with s...

      * 410039 : totem crashed with SIGSEGV in new_decoded_pad_full()

      * 340842 : do latency calculation for live sources

      * 341078 : RB does not play beyond initially downloaded podcast file

      * 414496 : [id3demux, id3v2mux] Add support for GST_TAG_MUSICBRAINZ_...

API additions since 0.10.12:

      * add gst_tag_freeform_string_to_utf8()

      * GstRTPBuffer::gst_rtp_buffer_default_clock_rate()

      * GstBaseAudioSink::slave-method property

      * add "min-ptime" property to RTP base audio payloader

      * gst_base_rtp_audio_payload_push()

      * gst_base_rtp_audio_payload_get_adapter()

      * GstMixerTrack::untranslated-label property

Changes since 0.10.11:

      * New API for on-demand plugin installation

      * Xv thread-safety and configuration enhancements

      * decodebin2 improvements

      * Support more raw audio format conversions

      * Improvements in Ogg support

      * AudioFilter base class ported to 0.10

      * Fixes for subtitles

      * Latency/live-playback support for Alsa

      * Lots of bug fixes and improvements

Bugs fixed since 0.10.11:

      * 398721 : No video in .ogm files with decodebin2

      * 339837 : [audioconvert] support for 64-bit float audio 

      * 341524 : [decodebin] can't handle decoders with always src pads wi...

      * 352069 : Add de.po German translation

      * 363379 : [oggmux] doesn't detect EOS on all sinkpads 

      * 378436 : [oggdemux] rhythmbox crash on fast clicking on rating in ...

      * 380342 : Totem does not play mp3 files when lyrics are present 

      * 383195 : [cddabasesrc,basertpaudiopayload] compile errors with gcc...

      * 383198 : totem crashed to gst_xvimagesink_update_colorbalance

      * 384008 : [xvimagesink] accesses - > xwindow outside locks

      * 384060 : gst_xoverlay_set_xwindow_id() causing lockups with x(v)im...

      * 387138 : x input events processing in sinks with xoverlay interfac...

      * 390063 : Documentation typo 

      * 390076 : add xv adaptor and port properties in xvimagesink element.

      * 391365 : [oggdemux] internal stream error on OggFlac

      * 392070 : [vorbis] GST_TAG_LOCATION not mapped

      * 392393 : [API] add libgstbaseutils library for missing plugins mes...

      * 396042 : mpeg4 video typefinder loops endlessly on quicktime redirect

      * 396835 : audioconvert/audioresample combination causing buffer of ...

      * 397673 : [patch] XIOError caught in x[v]imagesink.c

      * 397810 : [typefinding] .vob file: could not determine type of stream

      * 398110 : [theoraenc] GLib failed to allocate 3080991032 bytes on g...

      * 399340 : Crash in the oggdemux plugin when trying to play a specia...

      * 401029 : [playbin] rapidly changing visualisation freezes

      * 401072 : Move libgimme-codec helper functions to GStreamer

      * 402505 : visualisations don't work for some samplerates

      * 407811 : decodebin2 hang on HD clip

      * 409683 : Crash with Decodebin2

      * 410396 : not reading " DATE " tags from Flac files

      * 410963 : Fails to build with -z defs 

      * 357503 : [suparse] wrong timing with microdvd subtitles

      * 393310 : [pango] localtime_r does not exist in MinGW

      * 397207 : Test failure w/ HP-UX 11.11 & native compiler

      * 399948 : [textoverlay] leaks upstream events if textpad unlinked

      * 403963 : GstAudioFilter base class broken

      * 404512 : [videoscale] floating point exception on 1x1 video

      * 405020 : [alsa] probing the device-name doesn't seem to work corre...

      * 408278 : [videorate] memory leak

      * 410772 : Crash copying a GstNetBuffer

      * 401118 : [visual] error if width not a multiple of 4 

      * 405451 : [alsasink] deadlocks when disconnecting USB Sounddevice

API additions since 0.10.11:

      * GstAudioFilter

      * GST_VIDEO_SINK_CAST()

      * gst_pb_utils_add_codec_description_to_tag_list()

      * gst_pb_utils_get_codec_description()

      * gst_pb_utils_get_source_description()

      * gst_pb_utils_get_sink_description()

      * gst_pb_utils_get_decoder_description()

      * gst_pb_utils_get_encoder_description()

      * gst_pb_utils_get_element_description()

      * gst_pb_utils_init()

      * gst_install_plugins_context_new()

      * gst_install_plugins_context_set_xid()

      * gst_install_plugins_context_free()

      * gst_install_plugins_async()

      * gst_install_plugins_sync()

      * gst_install_plugins_return_get_name()

      * gst_install_plugins_installation_in_progress()

      * gst_missing_uri_source_message_new()

      * gst_missing_uri_sink_message_new

      * gst_missing_element_message_new

      * gst_missing_decoder_message_new

      * gst_missing_encoder_message_new

      * gst_missing_plugin_message_get_installer_detail

      * gst_missing_plugin_message_get_description

      * gst_is_missing_plugin_message

Bugs fixed since 0.10.10:

      * 360552 : [riff] [avi] extracts non-UTF8 metadata

      * 365501 : [x/xvimagesink] race condition when creating first image ...

      * 339366 : [playbin] hangs if suburi file type cannot be determined

      * 355914 : libvisual causes xvimagesink:  assertion `GST_CAPS_REFCOU...

      * 363118 : gst_riff_create_video_caps() should also store variant in...

      * 363607 : xvimagesink xwindow_draw_border() slowness

      * 336301 : [playbin] can't handle RTSP source

      * 337026 : oggmux doesn't set EOS properly

      * 337031 : vorbisdec outputs too much data 

      * 340049 : New BaseRTPAudioPayloader class to -base 

      * 348264 : Theora encoding, Ogg muxing don't handle discontinuities

      * 354773 : xvimage assumes that XV_COLORKEY can be set in RGB888 format

      * 355917 : libvisual plugin is broken

      * 355935 : multifdsink doesn't allow setting maximums (soft, hard) i...

      * 357038 : [ffmpegcolorspace] RGBA handling broken

      * 357215 : [playbin] buffering notification not quite right yet

      * 357289 : [riff] riff parser can't detect aac audio stream

      * 357404 : [playbin] Linking can fail silently 

      * 357531 : [subparse] problem if markup is not closed

      * 357577 : [playbin] regression: buffering still images broken

      * 357591 : Avoid compiler warning with uclibc and -Werror

      * 357613 : XvStopVideo in xvimagesink

      * 357800 : [libvisual] doesn't pass audio data to libvisual 0.4.0 co...

      * 359580 : tcpserversink and dataprotocol assert for multipart streams

      * 361095 : Fixes compiling with forte: warning clean up (part 3)

      * 361456 : [basertppayload] Memory leak

      * 361634 : sink- > ringbuffer NULL in BaseAudioSink's setcaps()

      * 361984 : [subparse] doesn't accept .srt file that doesn't start wi...

      * 366334 : [PATCH] Windows vs8 fixes

      * 368273 : Using the remove signal on multifdsink is not threadsafe

      * 368310 : include file  gstbasertpaudiopayload.h not included for r...

      * 369482 : [typefind] MPEG system streams get recognized as mp3 files

      * 370092 : [PATCH] Decodebin v2 : Implementation

      * 377183 : regression: no eos when playing ogg vorbis files

      * 381219 : bad debugging code left in audiorate

      * 382223 : [decodebin] more delayed linking

      * 382269 : Typefind detects mpeg video clip as audio/mpeg

      * 335635 : Add an Ogg/Vorbis retagging element

      * 341681 : [textoverlay] flickering with continuously timestamped text

      * 342228 : [alsa] Recognize " Front " as a Master channel 

      * 357330 : [subparse] some sami parser minor but enhanced patch 

      * 357532 : [gsttag] vorbistag doesn't handle dates that include time...

      * 359237 : [typefinding] doesn't recognize XML files shorter than 25...

      * 362845 : [subparse] add support for tmplayer format

      * 357977 : [videorate] new segment start is not respected

      * 364812 : [PATCH] oggmux release pad does not remove pad

      * 364856 : pngenc stride problems

      * 372507 : Mac build fixes

API added since 0.10.10:

      * playbin::queue-min-threshold property.

      * GstVideoOrientation interface

      * gst_base_rtp_depayload_push_ts

      * gst_base_rtp_depayload_push

      * Add dropped_buffers to multifdsink's get-stats GValueArray

      * gst_ring_buffer_commit_full

Changes since 0.10.9:

      * New elements: gdppay, gdpdepay

Bugs fixed since 0.10.9:

      * 343787 : The adder cannot handle when multiple elements tries to l...

      * 336075 : ALSA emu10k1 mixer tracks are wrongly classified as playb...

      * 349105 : crash with playbin and resizing screen

      * 342494 : [v4l] Query " device-name " even if device is not open

      * 342680 : [adder] seeking with multiple ogg files fails to work

      * 345188 : [alsa] can't handle more than 8 channels

      * 347091 : converting vorbis comments to GstTagLists is lossy

      * 348157 : Changed " Change Device " menu behaviour in gnome-volume-co...

      * 348916 : [typefind] add multipart/x-mixed-replace typefinder

      * 350157 : [riff] riff parser can't detect dts audio stream

      * 350655 : [oggdemux] should process seeking queries

      * 350900 : [adder] should not clamp floating point values

      * 351426 : API: add gst_tag_parse_extended_comment

      * 351502 : g_value_set_string leaks

      * 351742 : [vorbisenc] discontinuity detection too sensitive, might ...

      * 353658 : [videotestsrc] doesn't round strides correctly for YVYU

      * 354594 : multifdsink doesn't work reliably with sync-method = 'nex...

      * 351790 : [ogmparse] crash parsing video stream on x86-64

      * 140139 : [avidemux] can't play broken avi with ogg (not vorbis) au...

      * 347783 : [PLUGIN-MOVE] GDP elements should be moved

      * 347918 : Internal data flow error in udpsrc

      * 349656 : jitterbuffer in GstBaseRtp fails to handle rtp seqnum rol...

      * 350784 : element alsamixer doesn't respect asoundrc

      * 351308 : [netbuffer] build fails with gkt-doc critical warnings

      * 353234 : audiorate preserves DISCONT on buffers

      * 353912 : Add cmml caps to oggmux

API added since 0.10.9:

      * gst_rtp_buffer_get_payload_subbuffer()

      * gst_tag_parse_extended_comment()

      * GstPlayBin::connection-speed

      * GstTheoraParse::synchronization-points

      * GST_AUDIO_CHANNEL_POSITION_NONE

Changes since 0.10.8:

      * Parallel installability with 0.8.x series

      * Threadsafe design and API

      * Subtitle fixes

      * Support for images in tags

      * Playback improvements

      * Gnomevfssrc now supports burn:// uris

      * Videoscale now supports more RGBA formats

      * Multifdsink improvements

      * Testsuite can now generate coverage information

Bugs fixed since 0.10.8:

      * 347296 : Problems with clocks on alsasrc hangs the application

      * 347295 : [vorbisdec] Pushes before being initialized

      * 329798 : [playbin] doesn't always give correct error message for m...

      * 342085 : [alsasink] doesn't set buffer-time correctly

      * 342789 : [audioresample] doesn't clear state when stopped, causing...

      * 343303 : [subparse] workaround for bad entities in sami parser

      * 343385 : [gnomevfs] add support for burn:// URIs

      * 343500 : [riff] gst_riff_parse_strf_vids() can't parse extra data.

      * 343699 : oggmux leaks

      * 344503 : [subparse] parse font face property in sami parser.

      * 345131 : [PATCH] videoscale support for 32-bit RGB-formats

      * 345206 : [textoverlay] crash with non-UTF8 input

      * 345225 : [theoradec] Clipping for exact seeking

      * 345641 : [API] [libgsttag] add enums for image tag type

      * 345879 : [riff] won't play a .wmv file with WMVA video stream

      * 346581 : [typefinding] recognise text/html

      * 347221 : [audioconvert] channel remapping does not work right

      * 347304 : Massive leaks with xvimagesink

      * 346527 : alsasrc get_range does not respect requested size

Changes since 0.10.7:

      * alsasink probing fixes

      * xvimagesink error reporting fixes

      * subtitle fixes

      * adder fixes

      * vorbis multichannel fixes

      * multifdsink streamheader fixes

Bugs fixed since 0.10.7:

      * 169936 : [subparse] support for SAMI subtitles

      * 315312 : Gstreamer Xv uses RGB instead of YUV.

      * 334002 : video4linux shouldn't depend on X in configure script

      * 336881 : [libvisual] additional support for libvisual-0.4

      * 337544 : [xvimagesink] Internal Error when image is too large

      * 339520 : [subparse] add " encoding " property

      * 340909 : [alsasink] can't enable spdif output

      * 341542 : some users have an assertion failed: (GST_VIDEO_SINK_WIDT...

      * 341562 : audioconvert doesn't list formats in order of preference

      * 341696 : audioconvert crashes if converting from a format with no ...

      * 341719 : bisection algorithm in ogg doesn't bisect in some cases

      * 341732 : [alsasink] doesn't query supported sample rates

      * 341873 : [alsasink] minor memory leak, uses unprotected static var...

      * 342143 : [subparse] sami parser needs to escape characters

      * 342181 : [alsa] add property probe interface to alsasink and alsasrc

      * 342268 : [playbin] add 'subtitle-encoding' property

      * 342345 : [riff] Elephant's Dream AVI does not play, JUNK chunk bef...

      * 342566 : Building without GTK+ fails

      * 343397 : H.264/AAC movie deadlocks with totem in gstreamer code, p...

      * 339935 : [adder] dead-locks when adding sink pads in PAUSED state

Changes since 0.10.6:

      * typefind improvements

      * bug-fixes in textoverlay, audioconvert, videotestsrc, 

        multifdsink and audio source/sink base classes

      * Ice-cast metadata support has moved from gnomevfssrc to the 

        icydemux element in gst-plugins-good

      * audioresample now supports floating point samples

      * Adder element fixes.

      * Fixes for network playback and audio resampling in playbin

Bugs fixed since 0.10.6:

      * 340060 : [adder] handle newsegment events properly

      * 340375 : [API 0.11] [patch] typefind to differentiate between mp4 ...

      * 339405 : [textoverlay] can't display '\n' character

      * 338657 : [patch] adder should send events from src-pad to all sink...

      * 338919 : [patch] alsasink should also query witdh capabilities fro...

      * 301759 : [audioresample] float audio support (for OSX audio sinks)

      * 331901 : [videotestsrc] framerate=0/1 gives assertion error

      * 333657 : Replacing icy demuxing in gnomevfssrc

      * 336339 : [audioresample] should support width != 16

      * 338718 : [patch] [audioconvert] correctly clip float samples > 1.0

      * 338778 : [patch] Bad audio with ASX files

      * 338991 : [patch] Videoscale doesn't pass on pixel-aspect ratio

      * 339574 : [patch] Race condition in multifdsink can lead to spuriou...

      * 339786 : [typefinding] wavpack typefinding doesn't always work

      * 340369 : [volume element] " volume " property range insufficient

      * 340379 : [playbin] doesn't insert audioresample, causes problems w...

      * 340392 : Problem with internal-decodebin

      * 341160 : [multifdsink] client_status enum has an uninitialized nick

      * 341182 : Accessing playbin's streaminfo property from high languag...

      * 341432 : [playbin] automatically get icecast metadata requiring ic...

      * 341542 : some users have an assertion failed: (GST_VIDEO_SINK_WIDT...

      * 341557 : Map GST_TAG_IMAGE < = > ID3v2 APIC tag

API added since 0.10.6:

      * client-fd-removed signal added to multifdsink

      * stream-info-value-array property added to playbin

      * gst_video_calculate_display_ratio() in libgstvideo

Changes since 0.10.5:

      * QoS in sinks and transform elements

      * Needs GStreamer 0.10.5 for new GstBaseSink::async_playback() vmethod

      * added theoraparse element

Bugs fixed since 0.10.5:

      * 313136 : [playbin] hang while playing truncated ogg file

      * 172848 : [subparse] subtitles with special chars are displayed as ...

      * 305279 : [riff] uncompressed AVIs with 24bpp don't work

      * 320765 : [ffmpegcolorspace] make win32+msvc compliant, don't use _...

      * 323852 : Disable tests/icles on platforms that do not have X

      * 325653 : build errors compiling audioresample on win32(vs7)

      * 327357 : gst-plugins-base fails to compile with GCC 4.1

      * 334620 : [gnomevfssrc] fails to connect to icecast streaming servers

      * 334822 : [ffmpegcolorspace] YVU9 support

      * 335028 : [typefinding] ID3 v1 tag is not recognized with mp3-in-wa...

      * 335365 : inefficient use of GList in gst-plugins-base

      * 336190 : [gnomevfssink] should accept non-URI filenames as " location " 

      * 336194 : [gnomevfssrc] some minor memory leaks

      * 336477 : plugins need better/univied descriptions

      * 336617 : Unable to recognise MPEG TS stream

      * 337548 : Memory leaks in basertpdepayload

      * 337945 : [oggdemux] segment stop position ignored

      * 338419 : Regression in the handling of files with multiple audio/s...

      * 338897 : Videoscale crashes as part of DVD to Ogg transcoding

      * 339013 : [videorate] Goes into an infinite loop

      * 339047 : [riff] handle H264 fourcc in addition to h264

      * 339212 : ISO file typefinding regression

      * 330748 : deadlock in base audio sink on playing- > paused state change

Bugs fixed since 0.10.4:

      * 334216 : [gnomevfssrc] won't open some media on NFS mounts any longer

      * 334226 : typefindfunctions plugin crashes on PPC on registration

Changes since 0.10.3:

      * (Experimental) QoS support

      * oggmuxer now creates 100% valid streams for Theora, Vorbis and Speex

      * documentation updates

      * better support for subtitles (seeking)

Bugs fixed since 0.10.3:

      * 310202 : [subtitles] < i >  < /i > tags and others should be supported i...

      * 312439 : XVideo output doesn't work on remote displays (probably r...

      * 321271 : audio output is truncated at EOS

      * 321650 : Can't decode this ogm file

      * 325732 : [oggdemux] problem when seeking to time less than 4s with...

      * 325972 : [typefinding] doesn't recognise this mp3

      * 326720 : [alsasink] doesn't support more than 2 channels anymore

      * 330711 : [ffmpegcolorspace] problems with palettized RGB (fencount...

      * 330789 : gstbaseaudiosink causes noise on seeking

      * 330888 : Fix build with gcc 2.95 (again)

      * 331295 : gnomevfssink doesn't respect umask when creating files

      * 331526 : 3GP type detection is too simple

      * 331678 : Decodebin is not reusable within a single pipeline (as in...

      * 331690 : playbin won't play my last.fm stream

      * 331763 : [alsamixer] unmute sets the volume to 100%

      * 331765 : [alsamixer] mixer applet slider doesn't want to move from...

      * 331903 : [videorate] doesnt handle input caps of framerate=0/1 sanely

      * 332778 : [ogmparse] " Already an existing pad " WARNING

      * 332964 : random crashes in mp3_type_find

      * 333254 : theora encoder does not set IN_CAPS flag properly

      * 333352 : [gnomevfssink] reports disk full as generic error

      * 333488 : Allow for palette < 256 colours in AVI files

      * 333510 : [PATCH] Fix gst_pad_new_from_template (gst_static_pad_tem...

      * 333545 : [riff] set depth on wma caps to make asfdemux and pitfdll...

      * 333663 : [patch] unref the result of gst_pad_get_parent

      * 333900 : [typefind] cannot play a particular mp3 file

      * 334112 : variable not initialized

      * 334129 : Disable frame dropping for now

      * 317038 : use default channel layout if none is specified in multic...

      * 319340 : [cdparanoia] uncorrected-error signal never fired

API added since 0.10.3:

      * GstTextOverlay::halignment

      * GstTextOverlay::valignment

Changes since 0.10.2:

      * typefind improvements

      * Ogg decoding and encoding fixes

      * Improved audio and video sink classes

      * Bug and leak fixes

      * Improved video scaling

      * On-the-fly visualisation switching

      * Subtitle support

Bugs fixed since 0.10.2:

      * 330244 : gsttextoverlay.c:895: 'struct _GstCollectData' has no mem...

      * 324000 : [playbin] post error or message on unknown input

      * 153004 : [typefind] can't identify mp3 file with one single mpeg f...

      * 323874 : [playbin] leaks sinks and threads when using gconfaudiosink

      * 324626 : ffmpegcolorspace support for fourcc " UYVY "

      * 326447 : check that all elements in -base pass queries they can't ...

      * 328263 : Fix build with gcc 2.95

      * 328279 : [decodebin] timeout issue when pre-rolling

      * 329326 : Fix oggmux removing pads from collect pads

Changes since 0.10.1:

      * ported gnomevfssink, cdparanoia

      * New library and base class: GstCddaBaseSrc

      * ported mixerutils.h

      * added 'sine-tab' waveform to audiotestsrc

      * added float audio to audiorate

Bugs fixed since 0.10.1:

      * 324216 : [cdparanoia] missing patches from 0.8

      * 324696 : [videotestsrc] does not start counting the time from zero...

      * 324900 : Problem compiling gst-plugins-base with Forte

      * 325984 : [playbin] cannot handle sources that produce raw audio/video

      * 325990 : patch videotestsrc for using glib types

      * 326601 : GstRingBuffer crashes with alaw/mulaw caps

      * 327114 : [theoradec] should post tags on the bus

      * 327216 : vorbisdec segfaults on certain queries

API added since 0.10.1:

      * added libgstcddabase

      * added mixerutils.h

Changes since 0.10.0:

      * Parallel installability with 0.8.x series

      * Threadsafe design and API

      * removed gst-launch-ext

      * Ported: ogmparse

      * Fixes for: subparse, xvimagesink, audioresample, videorate, decodebin

Bugs fixed since 0.10.0:

      * 322347 : GstBaseRtpDepayload timestamps are wring

      * 323900 : Basertpdepayloader lets NEWSEGMENT events through unfiltered

      * 323878 : missing < string.h > inclusion (for memset & FD_ZERO)

API added since 0.10.0:

      * GstAlsaMixer::device

      * GstAlsaMixer::device-name

Bugs fixed since 0.9.7:

      * 319172 : gstreamer-plugins-base-0.9.pc doesn't export linking flags

      * 323017 : While(1) loop with sleep(0) in basertpdepayload.c

Changes since 0.9.6:

      * Parallel installability with 0.8.x series

      * Threadsafe design and API

      * ximagesink and xvimagesink updates and interactive test

      * added pango

      * rename net to netbuffer library

      * rtp element renaming

      * stream selector fixes

Bugs fixed since 0.9.6:

      * 319618 : [decodebin] some ogg videos don't play

      * 320644 : RTP packetizer does't set the packet timestamps correctly

      * 322388 : xvimagesink force-aspect-ratio=True always displays squar...

      * 322704 : oggdemux typefind list leak

Changes since 0.9.5:

      * Parallel installability with 0.8.x series

      * Threadsafe design and API

      * lots of leak fixes

      * flicker-free and rewritten X sinks

      * fractional framerates

      * removed sinesrc, replaced by audiotestsrc

Bugs fixed since 0.9.5:

      * 316442 : playbin should use autoaudiosink/autovideosink by default

      * 318353 : [ffmpegcolorspace] forward-port fixes from 0.8 branch

      * 320200 : vorbisenc: min-bitrate and max-bitrate are 1/1000 bps rat...

      * 321164 : gstringbuffer stops working under load

      * 321426 : ximage plugin should be renamed to ximagesink

      * 321446 : sinesrc should be dropped in favour of audiotestsrc

      * 321451 : GstRtpBuffer: no way to create a sub buffer with only the...

      * 321816 : [API] xoverlay API to post prepare-xwindow-id message

      * 321894 : vorbisenc doesn't compile

      * 322117 : Rename libgsttagedit to libgsttag

Changes since 0.9.4:

      * video caps now use a good range for framerate and w/h

      * oggdemux/oggmux improvements

      * playbin improvements

Bugs fixed since 0.9.4:

      * 319110 : [PATCH] oggdemux chain finding is slow

      * 320058 : playbin of a jpeg over http does not work

      * 320923 : [volume] doesn't build on Solaris

      * 321011 : gstbasertpdepayload doesn't send the " new segment " event ...

Changes since 0.9.3:

      * New element: audiotestsrc

      * typefind improvements

      * buffer-frames removed

Changes since 0.9.2:

      * RTP base classes

Bugs fixed since 0.9.2:

      * 313251 : ximagesink unused functions

      * 315159 : audioconvert lost 24 bit conversions in the rewrite