1 This is GStreamer Base Plug-ins 0.10.28, "Those Norwegians"
6 * build: really dist qtgv-xoverlay.h header file needed by overlay examples this time
7 * rtspconnection: fix handling of x-server-ip-address
10 Bugs fixed since 0.10.27:
12 * 610832 : qtgv-xoverlay.h header file missing in the tarball
13 * 611900 : [oggdemux] Incorrect parsing of Dirac headers
15 Changes since 0.10.26:
17 * playbin2,decodebin2: lots of fixes for missing plugin installation
18 * playbin2, playsink, subtitleoverlay: Set subtitle encoding properly
19 * videorate: Improve upstream negotiation
20 * oggdemux: use the chain begin_time instead of our counter
21 * oggdemux: mark skeleton streams correctly
22 * oggdemux: theora PAR of 0:N, N:0 or 0:0 is allowed and maps to 1:1
23 * typefinding: detect stm module format
24 * ffmpegcolorspace: add conversions from all ARGB formats to AYUV and back
25 * theoradec: Fix chroma copying for 4:2:2
26 * tcpclientsrc,tcpserversrc: Fix handling of closed sockets
27 * examples,build: dist header file for the Qt graphics view example
28 * playsink: Reset the sink's state to NULL before unreffing it unless it's the same instance again
29 * rtspconnection: make sure not to dereference NULL username or password
30 * appsrc: Update segment duration and post a duration message if the duration changes
31 * vorbisdec: also support ivorbis tremor decoder
32 * rtsp: fail gracefully on bad Content-Length headers
33 * rtsp: ignore \n and \r as the first line
35 Bugs fixed since 0.10.26:
37 * 610449 : codec autodetection does not always work
38 * 608025 : [videorate] fails at upstream negotiation
39 * 608309 : [appsrc] Should request new data before the queue is empty
40 * 608417 : rtspsrc problem with \n and \r as first line
41 * 609063 : [vorbisdec] also support integer vorbis decoder (tremor) library implementation
42 * 609314 : typefind: Typefind does not handle .stm module format
43 * 609423 : [appsrc] gst_app_src_set_size() should update duration and post a duration message
44 * 610005 : [oggdemux] regression: bad seek granularity
45 * 610268 : [rtsp] NULL pointer reference in gstrtspconnection
46 * 610310 : [playbin2] Subtitle encoding property has no effect
47 * 610329 : [theoradec] doesn't copy all chroma lines for 4:2:2
48 * 610379 : [playbin2] doesn't play if text flag is unset and media has text subtitles
49 * 610386 : [tcpserversrc] Doesn't send EOS when socket is closed
50 * 610672 : overlay examples are now inconsistent and broken
51 * 610832 : missing header file in the tarball
52 * 611225 : [oggdemux] doesn't preroll big_buck_bunny_427x240.indexed.ogg in push mode
53 * 611227 : [oggdemux] no duration or seeking in local big_buck_bunny_427x240.indexed.ogg in pull mode
54 * 604131 : Totem can no longer open Matroska files that hold ASS subtitles
56 API added since 0.10.26:
58 * appsrc::min-percent property
61 Changes since 0.10.25:
63 * playbin2: make about-to-finish signal work for raw sources (e.g. audio CDs)
64 * playbin2: fix handling of the native audio/video flags
65 * playbin2: add flag to enable decodebin buffering
66 * playbin2: make subtitle error handling more robust and ignore late errors
67 * playbin2: improve subtitle passthrough in uridecodebin
68 * playbin2: new subtitleoverlay element for generic subtitle overlaying
69 * playbin2: proxy notify::volume and notify::mute from the volume/mute
70 elements (or audio sink)
71 * playbin2: don't stop completely on initialization errors from subtitle
72 elements; instead disable the subtitles and play the other
74 * decodebin2: rewrite autoplugging and how groups of pads are exposed
75 * uridecodebin: add use-buffering property that will perform buffering on
76 parsed or demuxed media.
77 * GstXOverlay: flesh out docs and add example for use with Gtk+ >= 2.18
78 * libgsttag: add utility functions for ISO-639 language codes and tags
79 * oggdemux: use internal granulepos<->timestamp mapper and make oggdemux
80 more like a 'normal' demuxer that outputs timestamps
81 * oggdemux: seeking improvements
82 * subparse: add qttext support
83 * ffmpegcolorspace: prefer transforming alpha formats to alpha formats
84 and the other way around
85 * libgstvideo: add functions to create/parse still frame events.
86 * theoraenc: make the default quality property 48.
87 * videotestsrc: add pattern with out-of-gamut colors
88 * theora: port to 'new' theora 1.0 API; make misc. existing properties
89 have no effect (quick, keyframe-mindistance, noise-sensitivity,
90 sharpness, keyframe_threshold); those either never worked or
91 aren't needed/provided/useful any longer with the newer API
92 * typefinding: misc. performance improvements and fixes
93 * baseaudiosink: make drift tolerance configurable
95 Bugs fixed since 0.10.25:
97 * 507131 : GStreamer does not play short ogg sounds
98 * 583376 : [typefind] Detects MP3 as h264
99 * 344013 : [oggdemux] use parsers to suck less
100 * 598114 : build overwrites interfaces/interfaces-enumtypes.h with wrong enumtypes
101 * 344706 : [playbin] problem changing subtitles and language
102 * 350748 : [ffmpegcolorspace] ffmpeg colorspace should prefer RGBA over RGB
103 * 499181 : audiorate inserting samples (due to rounding errors ?)
104 * 524771 : Can't seek in YouTube videos
105 * 537050 : [playbin2] QOS event problems
106 * 542758 : [playbin2] Hangs in PLAYING forever if caps are not a subset of pad template caps
107 * 549254 : [playbin/decodebin] Doesn't handle pads that are added much later than the other(s) correctly
108 * 563828 : [decodebin2] Complains about loops in the graph when demuxer output requires another demuxer
109 * 568014 : oggdemux/theoradec doesn't play last video frame
110 * 570753 : [playbin] Support subtitle renderers additional to subtitle parsers
111 * 574289 : [decodebin2] race in state change to PAUSED
112 * 577326 : tcpclientsrc stops working if set to PLAYING, PAUSED and PLAYING again
113 * 579394 : [playbin2] deadlock with wavpack files: type_found - > analyze_new_pad - > no_more_pads
114 * 584441 : [playbin2] if suburi preroll fails with error, playback should continue
115 * 584987 : [playbin2] [gapless] Fire a track-changed message on track change.
116 * 585681 : Subtitle selector doesn't work
117 * 585969 : [playbin2] [gapless] Position/Duration information mismatch on track change
118 * 587704 : " GstDecodeBin2: This appears to be a text file " error when playing files from a samba share
119 * 591625 : [alsasrc] odd timestamping on start
120 * 591662 : [playbin2] can't handle both text subtitles and subpictures
121 * 591677 : Easy codec installation is not working
122 * 591706 : [playbin2] Support of files with subtitle subpicture streams
123 * 594729 : theora: Convert to libtheora 1.0 API
124 * 595123 : [playbin2] Should hide the difference between subtitles and subpictures
125 * 595401 : gobject assertion and null access to volume instance in playbin
126 * 595427 : avoid x event thread if not needed
127 * 595849 : Fix Y41B strides in videotestsrc and gstvideo
128 * 596159 : rtspsrc hangs when connecting over http tunneled rtsp
129 * 596694 : [typefind] Detects quicktime as mp3
130 * 596774 : Speed up subtitle display after seek/switch
131 * 596981 : [audioresample] Compilation failure due to warning about use of %lu for guint64 variable
132 * 597537 : [streamvolume.c]The cube root function is not defined in Microsoft's CRT
133 * 597539 : [gststrpconnection.c] 'close' is not defined in Microsoft's CRT
134 * 597786 : [tag] enhance gst_tag_freeform_string_to_utf8 to handle 16-bit Unicode
135 * 598288 : [decodebin2] Plays a wav file but issues an error
136 * 598533 : [decodebin2] Post element message with the stream topology on the bus
137 * 598936 : DKS subtitle format
138 * 599105 : [baseaudiosink] Remove pulsesink < 0.10.17 hack after gst-plugins-good release
139 * 599154 : RtpAudioPayload can send out buffers that are not exact multiple of the frame size
140 * 599266 : Requires restart after installing codecs
141 * 599471 : uridecodebin: Store unused decodebin2 instances for further usage.
142 * 599649 : Support for frame-based subtitles using playbin2 and subparse
143 * 600027 : [playbin2,playsink] Should notify about volume/mute changes
144 * 600370 : [subtitleoverlay] New element to overlay video with subtitles in every supported format
145 * 600469 : gdpdepay: Clear adapter on flush and state change
146 * 600479 : Deadlock when playing movie with subtitles
147 * 600726 : [queue2] implement buffering-left argument to buffer messages
148 * 600787 : playbin2 has a problem with Ogg stream with " info "
149 * 600945 : silence buffers at start reusing pulsesrc
150 * 600948 : [uridecodebin] Improve all raw caps detection on pads
151 * 601104 : [cddabasesrc] always plays first track if device is specified
152 * 601627 : theoradec breaks timestamps
153 * 601772 : gst-rtsp-server crashing : bug fixed
154 * 601809 : seek example doesn't work with csw
155 * 601942 : Add a still-frame event to libgstvideo
156 * 602000 : [playbin2] [gapless] Does state change PLAYING- > PAUSED- > PLAYING while it should stay in PLAYING
157 * 602225 : Can't play another movie after using subtitles
158 * 602790 : New oggdemux parsers break theora/vorbis playback
159 * 602834 : [ffmpegcolorspace] does un-necessary conversion from RGB to ARGB
160 * 602924 : Text subtitle rendering regression
161 * 602954 : [oggdemux] can't get first chain on ogg/theora stream
162 * 603345 : [playbin2] textoverlay refcount issues in git
163 * 603357 : [subparse] support for QTtext
164 * 605100 : GNOME Goal: Remove deprecated glib symbols
165 * 605219 : Freezes nearly always when switching Audio CDs
166 * 605960 : new examples require GTK 2.18
167 * 606050 : Implement ptime support
168 * 606163 : textoverlay: Ignore zero framerate
169 * 606687 : playbin2: can't see video after setting native flags
170 * 606744 : Totem fails to play video file: " Can't display both text subtitles and subpictures. "
171 * 606926 : Vorbis: Implement Proper Channel Orderings for 6.1 and 7.1 Configurations
172 * 607116 : [playbin2] no 'about-to-finish' signal with audio CDs
173 * 607226 : Disallow setting the playbin uri property in state > = PAUSED
174 * 607381 : GST_FRAMES_TO_CLOCK_TIME() GST_CLOCK_TIME_TO_FRAMES() should round result
175 * 607403 : rtpaudiopayload: ptime is in milli-seconds, convert to nanosecs
176 * 607569 : Playing a chained ogg stream from HTTP pauses or freezes between songs
177 * 607652 : segfault with an ogg annodex file
178 * 607848 : typefind wrong classifies mp4 file as mp3
179 * 607870 : [oggdemux] OGM parsing broken
180 * 607926 : [oggdemux] regression with certain chained ogg stream
181 * 607929 : [oggdemux] regression: headers pushed twice at the beginnign of each stream
182 * 608167 : [decodebin2] Doesn't push out full topology
183 * 608179 : caps filter appearing after adder results in deadlock
184 * 608446 : [playbin2] post an error message if no URI is set
185 * 608484 : [playbin2] problem with redirect and reset to READY
186 * 608699 : [oggdemux] memory leak while demuxing
187 * 609252 : [theoradec] Doesn't handle unknown pixel aspect ratio properly
188 * 596078 : Playbin2 takes ref of audio-/video-sink parameter
189 * 596183 : decodebin2: Rewrite autoplugging and how groups of pads are handled
190 * 601480 : [playback] Update factory lists not only after going back to NULL
191 * 596313 : gstv4lelement.c:168: error: ‘client’ may be used uninitialized in this function
192 * 606949 : [playbin2] verify type of volume property before using it
194 API added since 0.10.25:
196 * gst_rtcp_sdes_name_to_type()
197 * gst_rtcp_sdes_type_to_name()
198 * gst_tag_get_language_name()
199 * gst_tag_get_language_codes()
200 * gst_tag_get_language_code_iso_639_1()
201 * gst_tag_get_language_code_iso_639_2B()
202 * gst_tag_get_language_code_iso_639_2T()
203 * gst_video_event_new_still_frame()
204 * gst_video_event_parse_still_frame()
206 Changes since 0.10.24:
208 * Add per-stream volume controls
209 * Theora 1.0 and Y444 and Y42B format support
210 * Improve audio capture timing
211 * GObject introspection support
212 * Improve audio output startup
214 * Use pango-cairo instead of pangoft2
215 * Allow cdda://(device#)?track URI scheme in cddabasesrc
216 * Support interlaced content in videoscale and ffmpegcolorspacee
217 * Many other bug fixes and improvements
219 Bugs fixed since 0.10.24:
221 * 595401 : gobject assertion and null access to volume instance in playbin
222 * 563828 : [decodebin2] Complains about loops in the graph when demuxer output requires another demuxer
223 * 591677 : Easy codec installation is not working
224 * 588523 : smarter sink selection in playbin2
225 * 590146 : adder regressions
226 * 321532 : [cddabasesrc] Support device setting in cdda:// URI
227 * 340887 : add pangocairo textoverlay plugin.
228 * 397419 : [oggdemux] ogm video with subtitles stuck on first frame
229 * 556537 : [PATCH] typefind: more flexible MPEG4 start code recognition
230 * 559049 : gstcheck.c:76:F:general:test_state_changes_* failure: GST_IS_CLOCK(clock) assertion fails
231 * 567660 : [API] need a stream volume interface for sinks that do volume control
232 * 567928 : Make videorate work with a live source
233 * 571610 : [playbin] Scale of volume property is not documented
234 * 583255 : [playbin2] deadlock when disabling visualisations
235 * 586180 : RTSP improvements
236 * 588717 : [oggmux] gst_caps_unref() warning if not linked downstream
237 * 588761 : [videoscale] Needs special support for interlaced content
238 * 588915 : audioresample's output offset counter's initialization could maybe be improved
239 * 589095 : [appsrc] clarify documentation on caps and linkage
240 * 589574 : [typefind] incorrect sdp file detection
241 * 590243 : [videoscale] Claims to support MAX width/height
242 * 590425 : Slaved alsasrc clock with slave-method=re-timestamp not usable for RTP audio
243 * 590856 : [decodebin2] triggers assertion failure on NULL caps
244 * 591207 : totem does display the following subtitle srt file.
245 * 591357 : gst-plugins-base git won't build due to warning in gstrtspconnection.c
246 * 591577 : [playbin2] Incorrect error message string
247 * 591664 : [playbin2] after seeking, srt subtitles don't resync correctly
248 * 591934 : timestamp drift in audioresample
249 * 592544 : Remove regex.h check
250 * 592657 : [appsink] Blocks after entering on pause state
251 * 592864 : deadlocks from recent inputselector/streamselector change
252 * 592884 : [playbin2] g_object_get increases refcount by 2 and therefore leaves memleak
253 * 593035 : gdp doesn't preserve fields of the buffers put into the caps' streamheader
254 * 593284 : basertppayloader takes time in instance init
255 * 594020 : Totem don't play videos from ssh remote host
256 * 594094 : Playback Error playing Midi file
257 * 594136 : [alsasink] Regression from 0.10.23 -- element reuse doesn't work
258 * 594165 : [theoraenc] Implement support for new formats
259 * 594256 : improved slave-skew resynch mechanism
260 * 594258 : missing break in rtcpbuffer
261 * 594275 : Add cast to navigation to fix compiler warning
262 * 594623 : Expose playsink as a fully-fledged element
263 * 594732 : parse error
264 * 594757 : build fails due to warning in gstbasertppayload.c
265 * 594993 : [introspection] pkg-config file madness
266 * 594994 : [streamvolume] Add get_type function to the documentation
267 * 595454 : [cddabasesrc] uri format change breaks rhythmbox
268 * 545807 : [baseaudiosink] audible crack when starting the pipeline
270 API added since 0.10.24:
272 * gst_rtsp_connection_create_from_fd()
273 * gst_rtsp_connection_set_http_mode()
274 * gst_rtsp_watch_write_data()
275 * gst_rtsp_watch_send_message()
276 * GstBaseRTPPayload::perfect-rtptime
277 * GstBaseRTPAudioPayload::gst_base_rtp_audio_payload_flush()
278 * GstVideoSinkClass::show_frame()
279 * GstVideoSink:show-preroll-frame
280 * GST_MIXER_TRACK_READONLY
281 * GST_MIXER_TRACK_WRITEONLY
282 * GstStreamVolume interface
284 Changes since 0.10.23:
286 * Recognise Kate subpicture subtitles
287 * Support progressive download in playbin2
289 * Add buffer-list support in appsink
290 * Add gaussian-noise mode to audiotestsrc
291 * bump cdparanoia req to 0.10.2 and improve caching
292 * Improve audio source base class
293 * Add frame-by-frame stepping and examples
294 * Extend stream-probing in decodebin2
295 * Many RTSP improvements
296 * support for PGS subpictures
298 * Add Y444, v210, v216 formats
299 * implement preset interface in vorbisenc, theoraenc, oggmux
300 * Improve libvisual visualisation timestamp tracking
301 * playbin2 enhancements: custom audiosink, subpictures, cdda
302 * Improvements in textrender
303 * Support raw YUV 4:2:2 and SIREN in RIFF
304 * Add 4:2:2 and 4:4:4 support to theoradec
305 * Many other bug-fixes and improvements
307 Bugs fixed since 0.10.23:
309 * 510417 : [gio] make non-experimental
310 * 513373 : [PATCH] [gstvorbistag] Preserve cover art in Ogg/Vorbis tags
311 * 529300 : [giosink] [PATCH] Allow overwrite
312 * 531035 : [cdparanoia] Should depend on LGPL'd version of the libra...
313 * 567997 : [patch] add allow-pull-scheduling property to audio sinks
314 * 576552 : [subparse] post GST_TAG_SUBTITLE_CODEC tags
315 * 577637 : [playbin2] expose temp-location property
316 * 579692 : mp3_type_find is over-optimistic
317 * 580318 : [tagdemux] drops tag events from upstream
318 * 581460 : [baseaudiosrc] Reusing audio source leads to null timesta...
319 * 581571 : ARGB and alignment added to textrender
320 * 582021 : autogen: libtoolize must be called before aclocal
321 * 582749 : uridecodebin caps property not implemented yet
322 * 582819 : multifdsink: add num-fds property
323 * 583867 : gdpdepay + identity cause failed assertions
324 * 584020 : [playbin2] inadvertently resets configured audio/video sinks
325 * 584686 : [playbin2] Need {audio,video,text}-tags-changed signals
326 * 585197 : [subparse] fails to detect subrip subtitles with fewer th...
327 * 585758 : Remove deprecated GTK+ symbols
328 * 585970 : gst_audioringbuffer_get_type is not thread safe
329 * 585994 : gst-rtsp-message doesn't support " Timestamp " filed
330 * 586331 : [cdparanoia] expose cd cache size parameter
331 * 586356 : [playbin2] use private copy of input-selector as long as ...
332 * 586519 : white Gaussian noise would be useful in audiotestsrc
333 * 587080 : rtsp fails to compile - doesn't see some ws2tcpip functions
334 * 587278 : Support for GstBufferList in appsink
335 * 587676 : Call tzset() before localtime_r(), in e.g. gst-plugins-ba...
336 * 587695 : Patches to add stream-status messages audio elements
337 * 587896 : " No stream given yet " error from giostreamsrc
338 * 587980 : gstchannelmix.c: protect debug code with GST_DISABLE_GST_...
339 * 588078 : [playbin2] Fails to go to READY again after an error
340 * 588205 : Pipeline with giostreamsrc will not enter playing state
341 * 588550 : build failure in git, missing gstinterfaces-0.10
342 * 588551 : queue2: download buffering fixes
343 * 588724 : [vorbisdec] empty encoder string causes GStreamer
344 * 588746 : [audiotestsrc] Make sure tags are properly serialized in ...
345 * 588747 : [adder] Serialize incoming in-band events (tags) in the d...
346 * 588748 : [adder] Check dataflow consistency in unit tests
347 * 589075 : [playbin2] changing volume doesn't work after stream rest...
348 * 589581 : typefinder: recognise more Kate subtitle categories
349 * 589622 : Cannot use both playbin and input-selector
350 * 589663 : gstreamer asserts in gstaudiofilter
351 * 589797 : alsasrc does not set GstAlsaSrc- > handle to NULL after snd...
352 * 590470 : [typefinding] certain flac-in-ogg files not detected any ...
353 * 536313 : [cdda] Remove sha1 copy once we depend on glib-2.16
354 * 579642 : [oggdemux] handle broken ogg/vorbis files better
355 * 582528 : playbin2 Audio CD playback broken since
356 * 583318 : Assertion from within playbin2
357 * 585079 : undefined references to gst_adapter_* functions in schro
358 * 585708 : [adder] Wrong handling of flushing seeks
359 * 588218 : Siren in .wav support
360 * 586920 : rtsp: needs < netinet/in.h > on FreeBSD
362 API added since 0.10.23:
364 * GstNetAddress::gst_netaddress_to_string()
365 * Add gst_rtsp_watch_queue_data()
366 * playbin2: Add {audio,video,text}-tags-changed signals
367 * Add gst_color_balance_get_balance_type()
368 * Add gst_mixer_get_mixer_type()
370 Changes since 0.10.22:
372 * New navigation API to support DVD playback
373 * playbin2 improvements
374 * RTSP extensions to allow extra headers and options
375 * Replace audioresampler with speexresample based code
376 * Support interlacing flags in the gstvideo library
377 * Support new RIFF formats
378 * Improve typefinding
379 * Support more frame formats in videoscale
380 * Many other bug-fixes and improvements
382 Bugs fixed since 0.10.22:
384 * 577637 : [playbin2] expose temp-location property
385 * 580120 : [playbin2] unit test fails
386 * 478512 : [alsamixer] volume control slider not working
387 * 574962 : rhythmbox crash in flac_type_find
388 * 564139 : Documentation of TCP plugins
389 * 577436 : xvimagesink should use xcontext- > depth and not count bits...
390 * 350311 : [playbin2] support for subpicture subtitles
391 * 378094 : Enable pango elements to handle UYVY
392 * 543591 : Gnonlin can not play theora streams
393 * 553295 : [riff] fuzzed AVI file causes segfault
394 * 565105 : Gstreamer does not change from READY back to PAUSED in sa...
395 * 565777 : [riff] unrecognised video fourcc 0x10000002 for mpeg2 in avi
396 * 566661 : [typefind] Fall back to file extension using uri query
397 * 567255 : [riff] doesn't detect codec_id 0x706d as AAC (amongst other)
398 * 567636 : [pbutils] Missing plugins code shouldn't ask for the same...
399 * 567740 : bogus warning in decodebin2?
400 * 568482 : linking problems in gst-plugins-base
401 * 569655 : [ffmpegcolorspace] Add UYVY422 to GRAY8 conversion function
402 * 570142 : Documentation is broken for uridecodebin
403 * 570356 : aac typefinder failure
404 * 570768 : [ximagesink] wrong mouse pointer position if output windo...
405 * 570832 : Add flags to enhance mixer interfaces
406 * 571009 : [tagdemux] WMA file with id3v2 tag causes assertion to fail
407 * 571147 : [ffmpegcolorspace/videotestsrc] Add support for packed/pl...
408 * 572577 : [playbin2] deadlock on shutdown
409 * 572872 : [ffmpegcolorspace] Add YVYU colorspace
410 * 572993 : [subparse] broken libregex dependency on Windows
411 * 573165 : Generate additional export files for gstreamer app plugin
412 * 573528 : Wrong format modifier in gstgiobasesink.c
413 * 573529 : In gstrtspconnection.c some functions are called with wro...
414 * 574293 : [decodebin2] deadlock on shutdown
415 * 574319 : Missing HAVE_PROCESS_H in win32/common/config.h
416 * 574447 : gstadder.c: line 904: error C2036: 'gpointer' : unknown size
417 * 574939 : [typefinding] flac typefinder mis-typefinds PDFs as flac ...
418 * 575550 : srt subtitle file keeps playbin2 from playing
419 * 575638 : kissfft copyright
420 * 575649 : [oggdemux] duration query in time format returns true wit...
421 * 576019 : On Windows queue2 can't write files longer than 2-4 GiB, ...
422 * 576142 : [vorbisenc] Non-header output buffers have NULL caps
423 * 576180 : [playbin2] Uses unref'd audiosink volume if using gconfau...
424 * 576586 : [alsamixer] gnome-sound-properties freeze
425 * 577054 : [videoscale] Not valgrind clean
426 * 577709 : Review new navigation API
427 * 577827 : [appsink] Have appsink new_buffer-callback return GstFlow...
428 * 578583 : [PATCH] multifdsink doesn't handle sync-method=latest-key...
429 * 578656 : Implement upstream GstForceKeyUnit events in theoraenc
430 * 579129 : pkgconfig: appsrc/appsink can not be linked to uninstalled
431 * 579130 : app: expose trivial type macros
432 * 579192 : gst_rtcp_packet_get_type should not assert on packet content
433 * 579203 : baseaudiosink: unparenting the ringbuffer in NULL causes ...
434 * 579267 : [rtspconnection] g_async_queue_new_full() is GLib-2.16 AP...
435 * 579463 : [cddabasesrc] [cdparanoiasrc] no longer emits discid
436 * 579668 : audioresample fails to build with --disable-gst-debug
437 * 579734 : [playbin] raw_decoding_mode seems to be set unconditionally
438 * 579912 : [decodebin2] multiqueue is too small in time (interleave ...
439 * 580470 : [audioresample] causes pipelines to go out of sync and be...
440 * 580952 : [audioresample] bad quality/pops compared to plughw
441 * 581727 : [playbin2] make playsink go to PAUSED async
442 * 569682 : playbin2 leaks request pad from input selector
443 * 580020 : [vorbisenc] causes buffers to be out of segment if new se...
444 * 562794 : rtspsrc fails to create a socket on Win32 sometimes.
445 * 567396 : playbin2: DECODE_BIN_LOCK occasionally called twice withi...
446 * 567982 : " queued_bytes " field isn't updated while flushing the que...
447 * 571299 : [appsink] Handoff callback API
448 * 574443 : rtsp win32 - forgotten variable
449 * 574516 : [typefind] add typefinder for photoshop .psd files
450 * 574964 : gst_app_src_end_of_stream(), mutex on error return
451 * 575256 : rtspsrc fails to resolve hostnames
452 * 575588 : decodebin2 deadlock
453 * 576187 : [playbin2] Stalls video sink when disabling subtitles in ...
454 * 576188 : [playbin2] Reusing a playbin2 instance with visualization...
455 * 576190 : [playbin2] Deadlock when reusing playbin2 after an error
456 * 577288 : " Internal playbin error " when seeking to the end of files
457 * 577610 : RTCP feedback messages support in GstRTCPPacket
458 * 577794 : [playbin2] leaks elements set through properties
459 * 578118 : [multifdsink] add option to not resend the streamheader w...
460 * 578506 : Pipeline with alsasrc and alsasink cannot change state ba...
461 * 578942 : Missing RTSP headers related to Windows Media extension.
462 * 580271 : videorate: fails to clear discont flag on duplicated buffers
463 * 580649 : uridecodebin: bug on documentation published in website
465 API added since 0.10.22:
467 * GstRTSP::gst_rtsp_options_as_text()
468 * GstRTSPMessage::gst_rtsp_message_take_header()
469 * GstRTSPRange::gst_rtsp_range_to_string()
470 * New Navigation interface commands, queries and messages
471 * gst_rtsp_channel_new()
472 * gst_rtsp_channel_unref()
473 * gst_rtsp_channel_attach()
474 * gst_rtsp_channel_queue_message()
475 * gst_rtsp_connection_accept()
476 * GstAppSink::gst_app_sink_set_callbacks()
477 * GST_VIDEO_FORMAT_YVYU,GST_VIDEO_BUFFER_TFF,GST_VIDEO_BUFFER_RFF,GST_VIDEO_BUFFER_ONEFIELD
478 * GST_MIXER_FLAG_HAS_WHITELIST,GST_MIXER_FLAG_GROUPING,GST_MIXER_TRACK_NO_RECORD,GST_MIXER_TRACK_NO_MUTE,GST_MIXER_TRACK_WHITELIST
479 * GstAppSrc::emit-signals
480 * GstAppSrc::gst_app_src_set_emit_signals()
481 * GstAppSrc::gst_app_src_get_emit_signals()
482 * GstAppSrc::gst_app_src_set_callbacks()
483 * RTSP::gst_rtsp_connection_get_url()
484 * GstRTSPLowerTrans::GST_RTSP_LOWER_TRANS_HTTP
485 * RTSP:gst_rtsp_connection_set_tunneled()
486 * RTSP:gst_rtsp_connection_is_tunneled()
487 * RTSP::gst_rtsp_connection_set_ip()
488 * RTSP::gst_rtsp_connection_get_tunnelid()
489 * RTSP::gst_rtsp_connection_do_tunnel()
490 * RTSP::gst_rtsp_watch_reset()
494 1) Please note that decodebin2 and playbin2 API included in this release is
495 still considered unstable and WILL change in future releases. At this stage,
496 only developers or early adopters should consider using decodebin2 or playbin2
497 API embodied in their signals and properties.
499 Changes since 0.10.21:
501 * Require gettext 0.17
502 * Replace audioresample with speexresample from -bad
503 * Support new formats in RIFF: uncompressed RGB, WMA lossless, VP6
504 * Move libgstapp and elements from -bad
505 * Support color-key setting and probing for Xv properties
506 * Improve typefinding for various formats
507 * Extend audio sinks for pull-mode operation
508 * Support for more subtitle formats
509 * More development on decode2bin and playbin2
511 * Many bug fixes and improvements
513 Bugs fixed since 0.10.21:
515 * 562163 : theoraenc likely ignoring segments
516 * 562258 : rtspsrc element takes long time to error out if the addre...
517 * 561789 : [volume] deadlocks with a controller attached
518 * 554533 : [xvimagesink] allow setting colorkey if possible
519 * 567511 : colorkey in xvimagesink gets reset when element is reused
520 * 116051 : libresample doesn't handle > factor of 2 rate conversion
521 * 346218 : [audioresample] doesn't do anti aliasing
522 * 385061 : [audioresample?] investigate high CPU usage
523 * 456788 : [subparse] can't handle UTF-16 charset encoded subtitle.
524 * 525807 : [vorbisenc] vorbisenc has problems with a gnlsource that ...
525 * 546955 : gstoggmux EOS handling issue
526 * 549417 : [audioresample] unit test fails on 64bit linux
527 * 549510 : audioresample doesn't negotiate ideal caps
528 * 552237 : UTF-16 srt confuses gstreamer, misdetected as mp3
529 * 552559 : Implementation of SLAVE_SKEW in baseaudiosrc
530 * 552569 : audioresample producing strange sized buffers
531 * 552801 : audioconvert can overflow with big audio buffers
532 * 554879 : Add ability to specify format for date/time display in Gs...
533 * 555257 : Doesn't display srt subtitles saved with BOM
534 * 555319 : add FFV1 fourcc to riff-media
535 * 555607 : subrip subtitles typefind too strict
536 * 555699 : [PATCH] theoradec: prefer container's pixel aspect ratio ...
537 * 556025 : build failure in tests/icles
538 * 556066 : Last byte of FLAC image buffer chopped off
539 * 557365 : subparse check fails
540 * 558124 : [PLUGIN-MOVE] Move speexresample as audioresample2 to -base
541 * 559111 : ALSA sink hangs on USB audio device unplug while playing
542 * 559478 : does not play windows media streams correctly
543 * 559567 : `gst_base_audio_sink_sync_latency' should call `gst_base_...
544 * 561436 : videorate element add image/jpeg to caps template
545 * 561734 : playbin2 additions
546 * 561780 : Playbin2 should work without volume too
547 * 561924 : oggdemux hangs when given corrupt input via non-seekable ...
548 * 562270 : build without gdk fails
549 * 563143 : ximagesink/xvimagesink : _alloc_buffer returns non-clean ...
550 * 563174 : Implement gst_rtcp_packet_remove
551 * 563508 : [rgvolume] Unit test fails with passthrough assertions
552 * 563718 : Theora check out of date
553 * 563904 : GNOME Goal: Clean up GLib and GTK+ includes
555 API added since 0.10.21:
557 * clockoverlay::time-format
558 * GstRingBuffer:gst_ring_buffer_activate()
559 * GstRingBuffer:gst_ring_buffer_is_active()
560 * GstRingBuffer:gst_ring_buffer_convert()
561 * Add GST_TYPE_BASE_AUDIO_(SRC|SINK)_SLAVE_METHOD to the public API
562 * gst_netaddress_get_address_bytes()
563 * gst_netaddress_set_address_bytes()
565 Changes since 0.10.20:
567 * Continue playbin2 development
568 * Ogg improvements - CELT support, skeleton fixes
569 * DVD subpicture support
570 * Improved audio dithering random number generator
571 * xvimagesink/ximagesink fixes
572 * Vorbis encoding and decoding fixes
573 * Recognise Kate subtitle streams
574 * Many bug-fixes and enhancements
576 Bugs fixed since 0.10.20:
578 * 537380 : [gnomevfssrc] Doesn't handle short reads properly
579 * 538656 : xvimagesink support for autofill/colorkey property
580 * 540334 : Build fails without X in tests/examples/seek
581 * 528299 : Multiple GstMixerTracks with the same label cause problem...
582 * 530068 : Ogg Streams with Skeleton and Granulepos > 0 do not work(...
583 * 537009 : playbin2 silly typo breaks signals
584 * 537045 : decodebin2 sometimes emits 'drained' multiple times
585 * 537599 : [oggdemux] skeleton streams not skipped in ogg
586 * 537889 : [xvimagesink] colorbalance is bad
587 * 538232 : vorbisenc/vorbisdec don't work with a live source
588 * 538663 : gdppay memleak in gst_gdp_pay_reset
589 * 540215 : decodebin does not insert a queue for raw data type
590 * 540351 : [avidemux] Doesn't know about Duck DK4 ADPCM
591 * 540497 : ffmpegcolorspace is returning wrong size
592 * 541358 : cross mingw32 gcc: getaddrinfo is not in ws2_32.dll befor...
593 * 544306 : rtspsrc debug=1 segfaults with some libc
594 * 548898 : GStreamer-CRITICAL errors on seeking beyond stream borders
595 * 548913 : vorbisenc being picky about rounding errors in timestamps
596 * 549062 : Video devices aren't updated on subsequent probing.
597 * 549814 : [typefind] add application/pdf typefinder
598 * 550582 : [oggdemux] KATE streams not recognised
599 * 550638 : [typefind] Recognize some jpeg2k file types
600 * 550656 : recognize TrueSpeech in wavparse
601 * 550729 : gst-plugins-base won't compile with " -pedantic " option
602 * 552960 : tagdemux asserts and aborts on truncated files
603 * 553244 : theoraparse doesn't work at all (throws criticals and ass...
605 API added since 0.10.20:
607 * Add "index" property to GstMixerTrack to differantiate between
608 multiple mixer tracks with the same label.
610 Changes since 0.10.19:
613 * Support digest auth for RTSP
614 * Additional documentation
615 * Support DSCP QoS in multifdsink
616 * Add NV12/NV21 video buffer layouts
617 * Video scaling now bilinear by default
618 * Support more than 8 channels in audio conversions
619 * Channel mapping fixes for audioconvert
620 * Improve tmplayer and sami subtitle support
621 * Support 1x1 pixel buffers for videoscale
622 * Typefinding improvements for MPEG2, musepack
623 * Ogg/Dirac mapping updated in oggmux
624 * Fixes in ogg demuxing
625 * audiosink synchronisation and slaving fixes
626 * Support muting of the audio in playbin by selecting -1 as the audio stream
627 * Work done on playbin2 and uridecodebin
628 * Improvements in the experimental GIO plugin
630 * Handle GAP buffers in some places
631 * Various other leak and bug-fixes
633 Bugs fixed since 0.10.20:
635 * 526794 : [giosrc] totem doesn't work with some gvfs backends
636 * 510417 : [PLUGIN-MOVE] Move gio to gst-plugins-base
637 * 509125 : crash in CD Player: - playing CD - lowering/...
638 * 517813 : [audioconvert] make gap aware
639 * 302798 : [playbin] add mute property
640 * 342294 : Setting playbin property current-audio=-1 also stops the ...
641 * 398033 : [audioconvert] support more than 8 channels
642 * 419351 : [avi/a52dec] AV synchronization problems
643 * 467911 : [subparse] sami parser update
644 * 469933 : multifdsink IPv6 and diffserv TOS/TC markup
645 * 506659 : [textoverlay] rendering error when using non-standard widths
646 * 512333 : [gstvorbistag] Retrieve Ogg/Vorbis cover art as image met...
647 * 512382 : [playbin] race condition when pausing/playing multiple in...
648 * 518037 : pbutils-enumtypes.c is not included in win32/vs6/libgstpb...
649 * 521761 : gstaudioclock frozen the clock value until reaches latest...
650 * 522401 : gdpdepay doesn't validate payload CRCs
651 * 523993 : playbin2 blocks after a while when listening to a radio s...
652 * 524724 : [PATCH] [baseaudiosrc] buffer-time and latency-time do no...
653 * 525665 : Crash on Ogg/Vorbis with chain=NULL
654 * 525915 : [streamheader] Unit test fails with " gst_adapter_peek: as...
655 * 526173 : [typefinding] fails to detect mpeg video stream whereas m...
656 * 529018 : gst_ogm_parse_stream_header creates fraction value with w...
657 * 529500 : [videotestsrc] support for NV12 and NV21
658 * 529546 : [Playbin] Memory leak in streaminfo handling
659 * 530068 : Ogg Streams with Skeleton and Granulepos > 0 do not work(...
660 * 530531 : [typefinding] bad read in mpeg_video_stream_type_find
661 * 530719 : gst_video_calculate_display_ratio fails when playing Ogg ...
662 * 530962 : [subparse] parses only every second line of TMPlayer subt...
663 * 532454 : [NV12/NV21] videotestsrc and ffmpegcolorspace don't play ...
664 * 533087 : GstRTSPTransport kept opaque in docs
665 * 533817 : [audioconvert] Can't use default 7 channel layout / only ...
666 * 534071 : Gdppay memleak
667 * 534331 : race in decodebin when changing states while the internal...
668 * 535356 : vorbisdec doesn't support 8 channels
669 * 536475 : gdppay memleak and possible crash
670 * 536521 : Refcounting errors in playbin
671 * 536874 : Build failure on windows
672 * 532166 : [ffmpegcolorspace] support NV12 format
673 * 533617 : [audioconvert] Produces silence when converting 1/2 chann...
674 * 536848 : [giosrc] Doesn't handle short reads properly
675 * 536849 : [giosrc] Very slow doing any playback
676 * 518082 : [alsamixer] playback volumes overwritten by capture volum...
677 * 435633 : [PATCH] videorate not (fully) segment aware; causes frame...
678 * 532364 : tcpclientsrc broken in 0.10.19
679 * 533075 : gst_rtp_buffer_compare_seqnum doesn't do what it says
680 * 533265 : [cddabasesrc] Sound Juicer cut a sector when ripping a track
682 API additions since 0.10.20:
684 * decodebin2::sink-caps property
685 * giosrc::file property
686 * giosink::file property
687 * gst_base_audio_src_set_slave_method()
688 * gst_base_audio_src_get_slave_method()
689 * GstAudioClock::gst_audio_clock_reset()
690 * GstBaseAudioSrc:actual-buffer-time property
691 * GstBaseAudioSrc:actual-latency-time property
692 * gst_audio_check_channel_positions()
693 * add gst_tag_image_data_to_image_buffer()
694 * add gst_tag_list_add_id3_image()
695 * add GST_TAG_IMAGE_TYPE_NONE enum value
697 Changes since 0.10.18:
699 * Handle EAGAIN when polling sockets in rtspconnection
701 Changes since 0.10.17:
703 * Experimental GIO plugin
704 * Continued playbin2 development
706 * Better network element support on Windows
707 * Various other bug-fixes and improvements
709 Bugs fixed since 0.10.17:
711 * 509637 : [API] [basertpaudiopayload] add _set_samplebits_options()
712 * 510229 : [gnomevfssrc] HTTPS support
713 * 511478 : [rtpbuffer] add gst_rtp_buffer_set_extension_data function
714 * 511810 : [RTSP] Uses MT-unsafe gmtime() function
715 * 512899 : [alsa] gstalsasink.c:527: warning: 'snd_pcm_sw_params_set...
716 * 513167 : Fix compiler warning due to disabled signals in mixertrac...
717 * 514307 : [playbin] warning in nautilus, volume element can't be cr...
718 * 514623 : Ogg Theora video slow
719 * 514937 : Correct initialization of hints in is_multicast_address()
720 * 515654 : xvimagesink doesn't build with --disable-xshm
721 * 516246 : [alsasink] handle negative delay from snd_pcm_delay
722 * 517420 : typefind: add h264 elementary stream discovery
723 * 517991 : problems with configure file depending on GCC compiler
724 * 518039 : libgstrtsp MSVC 6.0 compile error
725 * 518162 : [subparse] handle italic text starting with " / " with Micr...
726 * 518940 : [playbin2] make _get_*_tags() match vfuncs prototype in c...
727 * 519906 : [API] add GstMixerOptions::get_values vfunc
728 * 519916 : [API] add mixer-changed and options-list-changed messages
729 * 520523 : [API] Unreviewed changes to ringbuffer API
730 * 521743 : libgstnetbuffer.def exports not up to date
731 * 522625 : [video] gst_video_format_parse_caps() broken for RGBA for...
732 * 523054 : gstbasesrc crashes when called from typefind helpers
733 * 511825 : [RTSP] compiler warning on FreeBSD
734 * 520300 : [alsasrc] provide-clock=false messes up buffer durations
736 API added since 0.10.17:
738 * GstRTPBuffer:gst_rtp_buffer_set_extension_data()
739 * add GST_VIDEO_FORMAT_Y41B and GST_VIDEO_FORMAT_Y42B.
740 * add GstMixerOptions::get_values vfunc (#519906)
741 * add gst_mixer_options_list_changed(), gst_mixer_mixer_changed() and
742 gst_mixer_message_parse_options_list_changed(). Fixes #519916.
743 * gst_base_rtp_audio_payload_set_samplebits_options()
744 * GstNetBuffer::gst_netaddress_equal
746 Changes since 0.10.16:
748 * Work-around ABI breakage due to unfortunate use of the
749 GST_DISABLE_DEPRECATED macro
750 * Export 2 missing functions needed for bindings in the win32 build
751 * Initialise the GstRingBuffer GType from a thread-safe context
753 Bugs fixed since 0.10.16:
755 * 511825 : [RTSP] compiler warning on FreeBSD
756 * 513018 : crash in Volume Control: I typed my password at t...
757 * 512334 : g_critical() when using GstAudioFilter & GST_DEBUG
759 Changes since 0.10.15:
761 * Handle newer Theora granule-pos semantics
762 * Introducing first alpha version playbin2 - the upcoming successor to
764 * Fixes in playbin handling of stream-switching
765 * New API for uniform handling of raw-video format buffers.
766 * Improvements for RTSP/RTP handling
767 * RIFF lib additions for VC-1 and AVC1 fourccs
768 * Many other bug-fixes and improvements
770 Bugs fixed since 0.10.15:
772 * 506132 : Review of changes in video/video.h
773 * 320984 : [oggdemux] cannot handle multiple chains
774 * 373011 : [playbin] throws error when switching off subtitles
775 * 436756 : Intermittent crashes in Pidgin in audioclock g_type_class...
776 * 462740 : [streamselector] patch to improve default stream selection
777 * 486840 : [alsamixer] use _all variants when setting the mixer
778 * 497964 : theoraenc test fails
779 * 498228 : gst-plugins-base-0.10.15 does not compile on FreeBSD (Gen...
780 * 499697 : Provide better pkg-config files
781 * 502497 : [subparse] SubRip subtitles starting from 0 not recognised
782 * 503440 : The control sockets used by gstrtspconnection.c are never...
783 * 503930 : [cdda] warning: 'eos' may be used uninitialized in this f...
784 * 506928 : [alsamixer] add " PCM " as master fall back for cards that ...
785 * 508138 : [decodebin] does not error out if pad activation fails
786 * 509762 : missing file in win32/MANIFEST
787 * 511274 : gst_rtp_buffer_get_extension_data is returning FALSE when...
788 * 496731 : [PATCH] xvimagesink leaks memory if initialization fails
789 * 496761 : [PATCH] RTSP message leaks memory when uninitialized
790 * 500763 : SIGSEGV while playing ogg audio file
792 API additions since 0.10.15:
794 * New GstVideoFormat API and helper functions in libgstvideo
795 * gst_base_audio_sink_set_provide_clock()
796 * gst_base_audio_sink_get_provide_clock()
797 * gst_base_audio_sink_set_slave_method()
798 * gst_base_audio_sink_get_slave_method()
799 * gst_base_audio_src_set_provide_clock()
800 * gst_base_audio_src_get_provide_clock()
802 Changes since 0.10.14:
804 * RTP/RTSP/RTCP/SDP support improved
805 * New FFT support library libgstfft, based on Kiss FFT
806 * New formats supported in volume and audiotestsrc
807 * Fixes in audiorate and videorate
808 * Audio capture fixes
809 * Playbin and decodebin fixes
810 * New tagdemux base class for ID3/APE style tag readers
811 * Fix a nasty crash in the X sinks on shutdown
813 * Add support for multichannel WAV files.
814 * Preserve channel layout information when up/down-mixing.
815 * Many bug-fixes and improvements
817 Bugs fixed since 0.10.14:
819 * 475395 : decodebin2 leaks request-pads
820 * 475451 : [decodebin2] leaks ghostpad
821 * 378770 : [xvimagesink] race condition in event thread?
822 * 407282 : [decodebin2] autoplug-sort signal has GList ** parameter
823 * 430677 : [audioconvert] does not preserve channel positions when f...
824 * 442654 : [volume] controller bypassed by default
825 * 445529 : [volume] support for 24/32-bit audio/x-raw-int
826 * 446766 : return code for gst_base_rtp_payload_audio_handle_event()
827 * 451970 : Subparse requires HTML parser
828 * 453650 : [audiobasesrc] two alsasrcs do not work in one pipeline
829 * 459334 : [textoverlay] expose pango line alignment property
830 * 459585 : [basertpdepayload] api without namespace
831 * 460422 : [audiotestsrc] Add support for float and double output
832 * 462805 : [alsa] compilation fails with gcc 4.2
833 * 462979 : Add 'silent' property to GstTimeOverlay
834 * 463215 : [audioconvert] compile errors
835 * 464320 : [PATCH] gst-plugins-base-0.14 does not build for win32
836 * 464666 : [playbin] QT trailer hangs in preroll with decodebin2
837 * 464690 : Add connection-speed property to uridecodebin element
838 * 465015 : [playbin] Not removed probes causes deadlocks in streamin...
839 * 465028 : some warnings with mingw
840 * 467667 : GST_FRAMES_TO_CLOCK_TIME() and GST_CLOCK_TIME_TO_FRAMES()...
841 * 468129 : [basertpaudiopayload] event handler returns the wrong value
842 * 468619 : New library gstfft: FFT library for integer and float typ...
843 * 470456 : [API] add gst_missing_*_installer_detail_new()
844 * 470766 : [ssaparse] line breaks in SSA subtitle parser
845 * 471067 : Make the SDP code useable for generating SDP descriptions
846 * 471194 : [rtpbuffer] RTP headers are wrong for win32
847 * 473097 : [baseaudiosink] gstreamer-properties hangs when testing s...
848 * 474384 : gstrtsp-enumtypes.c and .h needed for win32
849 * 474880 : [xvimagesink] [ximagesink] leaking buffer caps reference
850 * 475731 : rtspconnection is able to read incomplete messages
851 * 483620 : All Rtp buffers are discarded -- gst_rtp_buffer_get_payl...
852 * 484989 : memleak, not unrefed caps for gstbasertppayload.c
853 * 489010 : Please change default channel order for WAVE_EXT-less .wa...
854 * 491722 : [playbin] regression: crash with external subtitles
855 * 492098 : [GstFFT] Broken scaling
856 * 492114 : Build issues on Windows/MSVC
857 * 492306 : compilation errors with MinGW
858 * 492813 : Missing symbols in libgstrtp.def
859 * 493986 : Build issues on Windows (missing symbols)
860 * 494346 : pre-release vs6 patch
861 * 496548 : Including malloc.h breaks macos build
862 * 496724 : DSW file references non-existent DSP files
863 * 464079 : audiotestsrc doesn't respond to conversion queries properly
864 * 442065 : floatcast.h includes config.h and might break other apps
865 * 466717 : gst_event_new_new_segment_full:assertion `start < = stop' ...
866 * 485753 : Decodebin2 deadlocks when nulling pipeline during typefind
867 * 464028 : Move connection-speed from playbin to playbasebin
869 API added since 0.10.14:
871 * GstTagDemux base class for simple tag demuxers
872 * GstBaseAudioSrc::provide-clock property
873 * gst_rtcp_ntp_to_unix()
874 * gst_rtcp_unix_to_ntp()
875 * gst_rtp_buffer_get_header_len()
876 * gst_rtp_buffer_get_extension_data()
877 * gst_rtp_buffer_compare_seqnum()
878 * gst_rtp_buffer_ext_timestamp()
879 * gst_rtcp_packet_sdes_copy_entry()
880 * gst_install_plugins_supported()
881 * gst_missing_*_installer_detail_new() convenience API
882 * gst_rtsp_connection_poll()
883 * GstTextOverlay::line-alignment property
885 Changes since 0.10.13:
887 * Audio dither and noise-shaping when reducing bit-depth
888 * RTSP and SDP helper libraries added
889 * Experimental buffering element "queue2" now supports pull-mode
890 and file-based buffering.
891 * Support for more 32-bit video pixel layouts
892 * Various fixes and improvements
894 Bugs fixed since 0.10.13:
896 * 380625 : [x*imagesink] add 'handle-expose' property
897 * 385527 : oggmux sometimes gets DELTA flag on output wrong near start
898 * 402076 : videoscale 4-tap method broken for downscaling
899 * 437169 : [xvimagesink] add property to disable Xv double-buffering
900 * 441264 : queue2 support to do buffering on a file
901 * 442553 : [v4lsrc] doesn't output segments in GST_FORMAT_TIME
902 * 442557 : [videorate] doesn't handle latency queries
903 * 442944 : Audiotestsrc can overflow on seeks
904 * 444523 : [queue2] Pull mode support
905 * 444630 : Compilation error with fsseko (from gstqueue2.c) -- unabl...
906 * 445505 : [queue2] It does not work in pull mode with oggdemux
907 * 446551 : [queue2] Buffering is not working properly if it is set t...
908 * 446572 : [queue2] Division by zero
909 * 446972 : warning when compiling gstoggdemux.c
910 * 449156 : Regression in CVS for decodebin2
911 * 450875 : Missing files in po/POTFILES.in
912 * 451707 : [tag] UTF-8 in ID3v1 tag not correctly decoded
913 * 451908 : [ffmpegcolorspace] regression: doesn't accept GST_VIDEO_C...
914 * 454264 : Playbin fails to " play " image url after a movie url
915 * 456656 : [API] Addition of audio buffer clipping function to gstaudio
916 * 460978 : gst_audio_buffer_clip outputs warnings
917 * 152864 : [PATCH] GstAlsaMixer doesn't support signals
918 * 360246 : [audioconvert] Optionally apply dithering
919 * 394061 : Add support for Subviewer subtitles
920 * 420326 : Base payloader class has wrong property types and ranges
921 * 451145 : [vorbisdec] errors out on 0-sized packets
922 * 459204 : [PATCH] [playbin] gst_play_base_bin_get_streaminfo_value_...
924 API added since 0.10.13:
926 * RTSP and SDP libraries added
927 * gst_rtsp_base64_decode_ip
928 * Add buffer clipping function gst_audio_buffer_clip for raw audio
929 buffers. Fixes #456656.
930 * gst_mixer_get_mixer_flags
931 * gst_mixer_message_parse_mute_toggled
932 * gst_mixer_message_parse_record_toggled
933 * gst_mixer_message_parse_volume_changed
934 * gst_mixer_message_parse_option_changed
935 * GstMixerMessageType
938 Changes since 0.10.12:
939 * Many fixes and improvements
940 * RTP and RTCP support improved
942 Bugs fixed since 0.10.12:
944 * 339838 : [audioconvert] support floats with non-native endianness
945 * 393975 : closing x/xvimagesink window crashes gst-launch
946 * 405072 : [API] add gst_tag_freeform_string_to_utf8()
947 * 413799 : [subparse] add support for MPL2 format
948 * 414645 : GstMixerTrack should make untranslated label available
949 * 420079 : [audioconvert] Uses biased rounding which results in dist...
950 * 420578 : [subparse] add more colour map in sami parser
951 * 421834 : videorate breaks on dimension changes
952 * 423051 : Vorbis tags of type double use locale-dependent formatting
953 * 423055 : Verify ReplayGain vorbistag processing in libs/tag testsuite
954 * 425455 : Decodebin2 leaks pads
955 * 426250 : GstPlayBaseBin leaks streaminfo objects
956 * 428187 : Rtp base depayloader class doesn't send new_segment after...
957 * 431672 : gst_base_rtp_audio_payload_push() should take object of i...
958 * 432362 : [ximagesink] doesn't build if XShm is not available
959 * 432755 : [videorate] leaks buffer if flow != OK
960 * 432984 : [baseaudiosrc] misleading warning message when dropping s...
961 * 433888 : [theoradec] does not generate a perfect stream
962 * 436562 : Theoradec doesn't work well with gnonlin
963 * 438840 : [theoradec] does not compile with old version of libtheora
964 * 440997 : [gstriff] Doesn't handle width!=depth files with audio/x-...
965 * 441295 : audioconvert doesn't build on VS6
966 * 442024 : regression in playbin buffering
967 * 350299 : [playbin] " Internal data flow error " opening movie with s...
968 * 410039 : totem crashed with SIGSEGV in new_decoded_pad_full()
969 * 340842 : do latency calculation for live sources
970 * 341078 : RB does not play beyond initially downloaded podcast file
971 * 414496 : [id3demux, id3v2mux] Add support for GST_TAG_MUSICBRAINZ_...
973 API additions since 0.10.12:
975 * add gst_tag_freeform_string_to_utf8()
976 * GstRTPBuffer::gst_rtp_buffer_default_clock_rate()
977 * GstBaseAudioSink::slave-method property
978 * add "min-ptime" property to RTP base audio payloader
979 * gst_base_rtp_audio_payload_push()
980 * gst_base_rtp_audio_payload_get_adapter()
981 * GstMixerTrack::untranslated-label property
983 Changes since 0.10.11:
985 * New API for on-demand plugin installation
986 * Xv thread-safety and configuration enhancements
987 * decodebin2 improvements
988 * Support more raw audio format conversions
989 * Improvements in Ogg support
990 * AudioFilter base class ported to 0.10
991 * Fixes for subtitles
992 * Latency/live-playback support for Alsa
993 * Lots of bug fixes and improvements
995 Bugs fixed since 0.10.11:
997 * 398721 : No video in .ogm files with decodebin2
998 * 339837 : [audioconvert] support for 64-bit float audio
999 * 341524 : [decodebin] can't handle decoders with always src pads wi...
1000 * 352069 : Add de.po German translation
1001 * 363379 : [oggmux] doesn't detect EOS on all sinkpads
1002 * 378436 : [oggdemux] rhythmbox crash on fast clicking on rating in ...
1003 * 380342 : Totem does not play mp3 files when lyrics are present
1004 * 383195 : [cddabasesrc,basertpaudiopayload] compile errors with gcc...
1005 * 383198 : totem crashed to gst_xvimagesink_update_colorbalance
1006 * 384008 : [xvimagesink] accesses - > xwindow outside locks
1007 * 384060 : gst_xoverlay_set_xwindow_id() causing lockups with x(v)im...
1008 * 387138 : x input events processing in sinks with xoverlay interfac...
1009 * 390063 : Documentation typo
1010 * 390076 : add xv adaptor and port properties in xvimagesink element.
1011 * 391365 : [oggdemux] internal stream error on OggFlac
1012 * 392070 : [vorbis] GST_TAG_LOCATION not mapped
1013 * 392393 : [API] add libgstbaseutils library for missing plugins mes...
1014 * 396042 : mpeg4 video typefinder loops endlessly on quicktime redirect
1015 * 396835 : audioconvert/audioresample combination causing buffer of ...
1016 * 397673 : [patch] XIOError caught in x[v]imagesink.c
1017 * 397810 : [typefinding] .vob file: could not determine type of stream
1018 * 398110 : [theoraenc] GLib failed to allocate 3080991032 bytes on g...
1019 * 399340 : Crash in the oggdemux plugin when trying to play a specia...
1020 * 401029 : [playbin] rapidly changing visualisation freezes
1021 * 401072 : Move libgimme-codec helper functions to GStreamer
1022 * 402505 : visualisations don't work for some samplerates
1023 * 407811 : decodebin2 hang on HD clip
1024 * 409683 : Crash with Decodebin2
1025 * 410396 : not reading " DATE " tags from Flac files
1026 * 410963 : Fails to build with -z defs
1027 * 357503 : [suparse] wrong timing with microdvd subtitles
1028 * 393310 : [pango] localtime_r does not exist in MinGW
1029 * 397207 : Test failure w/ HP-UX 11.11 & native compiler
1030 * 399948 : [textoverlay] leaks upstream events if textpad unlinked
1031 * 403963 : GstAudioFilter base class broken
1032 * 404512 : [videoscale] floating point exception on 1x1 video
1033 * 405020 : [alsa] probing the device-name doesn't seem to work corre...
1034 * 408278 : [videorate] memory leak
1035 * 410772 : Crash copying a GstNetBuffer
1036 * 401118 : [visual] error if width not a multiple of 4
1037 * 405451 : [alsasink] deadlocks when disconnecting USB Sounddevice
1039 API additions since 0.10.11:
1042 * GST_VIDEO_SINK_CAST()
1043 * gst_pb_utils_add_codec_description_to_tag_list()
1044 * gst_pb_utils_get_codec_description()
1045 * gst_pb_utils_get_source_description()
1046 * gst_pb_utils_get_sink_description()
1047 * gst_pb_utils_get_decoder_description()
1048 * gst_pb_utils_get_encoder_description()
1049 * gst_pb_utils_get_element_description()
1050 * gst_pb_utils_init()
1051 * gst_install_plugins_context_new()
1052 * gst_install_plugins_context_set_xid()
1053 * gst_install_plugins_context_free()
1054 * gst_install_plugins_async()
1055 * gst_install_plugins_sync()
1056 * gst_install_plugins_return_get_name()
1057 * gst_install_plugins_installation_in_progress()
1058 * gst_missing_uri_source_message_new()
1059 * gst_missing_uri_sink_message_new
1060 * gst_missing_element_message_new
1061 * gst_missing_decoder_message_new
1062 * gst_missing_encoder_message_new
1063 * gst_missing_plugin_message_get_installer_detail
1064 * gst_missing_plugin_message_get_description
1065 * gst_is_missing_plugin_message
1067 Bugs fixed since 0.10.10:
1069 * 360552 : [riff] [avi] extracts non-UTF8 metadata
1070 * 365501 : [x/xvimagesink] race condition when creating first image ...
1071 * 339366 : [playbin] hangs if suburi file type cannot be determined
1072 * 355914 : libvisual causes xvimagesink: assertion `GST_CAPS_REFCOU...
1073 * 363118 : gst_riff_create_video_caps() should also store variant in...
1074 * 363607 : xvimagesink xwindow_draw_border() slowness
1075 * 336301 : [playbin] can't handle RTSP source
1076 * 337026 : oggmux doesn't set EOS properly
1077 * 337031 : vorbisdec outputs too much data
1078 * 340049 : New BaseRTPAudioPayloader class to -base
1079 * 348264 : Theora encoding, Ogg muxing don't handle discontinuities
1080 * 354773 : xvimage assumes that XV_COLORKEY can be set in RGB888 format
1081 * 355917 : libvisual plugin is broken
1082 * 355935 : multifdsink doesn't allow setting maximums (soft, hard) i...
1083 * 357038 : [ffmpegcolorspace] RGBA handling broken
1084 * 357215 : [playbin] buffering notification not quite right yet
1085 * 357289 : [riff] riff parser can't detect aac audio stream
1086 * 357404 : [playbin] Linking can fail silently
1087 * 357531 : [subparse] problem if markup is not closed
1088 * 357577 : [playbin] regression: buffering still images broken
1089 * 357591 : Avoid compiler warning with uclibc and -Werror
1090 * 357613 : XvStopVideo in xvimagesink
1091 * 357800 : [libvisual] doesn't pass audio data to libvisual 0.4.0 co...
1092 * 359580 : tcpserversink and dataprotocol assert for multipart streams
1093 * 361095 : Fixes compiling with forte: warning clean up (part 3)
1094 * 361456 : [basertppayload] Memory leak
1095 * 361634 : sink- > ringbuffer NULL in BaseAudioSink's setcaps()
1096 * 361984 : [subparse] doesn't accept .srt file that doesn't start wi...
1097 * 366334 : [PATCH] Windows vs8 fixes
1098 * 368273 : Using the remove signal on multifdsink is not threadsafe
1099 * 368310 : include file gstbasertpaudiopayload.h not included for r...
1100 * 369482 : [typefind] MPEG system streams get recognized as mp3 files
1101 * 370092 : [PATCH] Decodebin v2 : Implementation
1102 * 377183 : regression: no eos when playing ogg vorbis files
1103 * 381219 : bad debugging code left in audiorate
1104 * 382223 : [decodebin] more delayed linking
1105 * 382269 : Typefind detects mpeg video clip as audio/mpeg
1106 * 335635 : Add an Ogg/Vorbis retagging element
1107 * 341681 : [textoverlay] flickering with continuously timestamped text
1108 * 342228 : [alsa] Recognize " Front " as a Master channel
1109 * 357330 : [subparse] some sami parser minor but enhanced patch
1110 * 357532 : [gsttag] vorbistag doesn't handle dates that include time...
1111 * 359237 : [typefinding] doesn't recognize XML files shorter than 25...
1112 * 362845 : [subparse] add support for tmplayer format
1113 * 357977 : [videorate] new segment start is not respected
1114 * 364812 : [PATCH] oggmux release pad does not remove pad
1115 * 364856 : pngenc stride problems
1116 * 372507 : Mac build fixes
1118 API added since 0.10.10:
1120 * playbin::queue-min-threshold property.
1121 * GstVideoOrientation interface
1122 * gst_base_rtp_depayload_push_ts
1123 * gst_base_rtp_depayload_push
1124 * Add dropped_buffers to multifdsink's get-stats GValueArray
1125 * gst_ring_buffer_commit_full
1127 Changes since 0.10.9:
1129 * New elements: gdppay, gdpdepay
1131 Bugs fixed since 0.10.9:
1133 * 343787 : The adder cannot handle when multiple elements tries to l...
1134 * 336075 : ALSA emu10k1 mixer tracks are wrongly classified as playb...
1135 * 349105 : crash with playbin and resizing screen
1136 * 342494 : [v4l] Query " device-name " even if device is not open
1137 * 342680 : [adder] seeking with multiple ogg files fails to work
1138 * 345188 : [alsa] can't handle more than 8 channels
1139 * 347091 : converting vorbis comments to GstTagLists is lossy
1140 * 348157 : Changed " Change Device " menu behaviour in gnome-volume-co...
1141 * 348916 : [typefind] add multipart/x-mixed-replace typefinder
1142 * 350157 : [riff] riff parser can't detect dts audio stream
1143 * 350655 : [oggdemux] should process seeking queries
1144 * 350900 : [adder] should not clamp floating point values
1145 * 351426 : API: add gst_tag_parse_extended_comment
1146 * 351502 : g_value_set_string leaks
1147 * 351742 : [vorbisenc] discontinuity detection too sensitive, might ...
1148 * 353658 : [videotestsrc] doesn't round strides correctly for YVYU
1149 * 354594 : multifdsink doesn't work reliably with sync-method = 'nex...
1150 * 351790 : [ogmparse] crash parsing video stream on x86-64
1151 * 140139 : [avidemux] can't play broken avi with ogg (not vorbis) au...
1152 * 347783 : [PLUGIN-MOVE] GDP elements should be moved
1153 * 347918 : Internal data flow error in udpsrc
1154 * 349656 : jitterbuffer in GstBaseRtp fails to handle rtp seqnum rol...
1155 * 350784 : element alsamixer doesn't respect asoundrc
1156 * 351308 : [netbuffer] build fails with gkt-doc critical warnings
1157 * 353234 : audiorate preserves DISCONT on buffers
1158 * 353912 : Add cmml caps to oggmux
1160 API added since 0.10.9:
1162 * gst_rtp_buffer_get_payload_subbuffer()
1163 * gst_tag_parse_extended_comment()
1164 * GstPlayBin::connection-speed
1165 * GstTheoraParse::synchronization-points
1166 * GST_AUDIO_CHANNEL_POSITION_NONE
1168 Changes since 0.10.8:
1170 * Parallel installability with 0.8.x series
1171 * Threadsafe design and API
1173 * Support for images in tags
1174 * Playback improvements
1175 * Gnomevfssrc now supports burn:// uris
1176 * Videoscale now supports more RGBA formats
1177 * Multifdsink improvements
1178 * Testsuite can now generate coverage information
1180 Bugs fixed since 0.10.8:
1182 * 347296 : Problems with clocks on alsasrc hangs the application
1183 * 347295 : [vorbisdec] Pushes before being initialized
1184 * 329798 : [playbin] doesn't always give correct error message for m...
1185 * 342085 : [alsasink] doesn't set buffer-time correctly
1186 * 342789 : [audioresample] doesn't clear state when stopped, causing...
1187 * 343303 : [subparse] workaround for bad entities in sami parser
1188 * 343385 : [gnomevfs] add support for burn:// URIs
1189 * 343500 : [riff] gst_riff_parse_strf_vids() can't parse extra data.
1190 * 343699 : oggmux leaks
1191 * 344503 : [subparse] parse font face property in sami parser.
1192 * 345131 : [PATCH] videoscale support for 32-bit RGB-formats
1193 * 345206 : [textoverlay] crash with non-UTF8 input
1194 * 345225 : [theoradec] Clipping for exact seeking
1195 * 345641 : [API] [libgsttag] add enums for image tag type
1196 * 345879 : [riff] won't play a .wmv file with WMVA video stream
1197 * 346581 : [typefinding] recognise text/html
1198 * 347221 : [audioconvert] channel remapping does not work right
1199 * 347304 : Massive leaks with xvimagesink
1200 * 346527 : alsasrc get_range does not respect requested size
1202 Changes since 0.10.7:
1204 * alsasink probing fixes
1205 * xvimagesink error reporting fixes
1208 * vorbis multichannel fixes
1209 * multifdsink streamheader fixes
1211 Bugs fixed since 0.10.7:
1213 * 169936 : [subparse] support for SAMI subtitles
1214 * 315312 : Gstreamer Xv uses RGB instead of YUV.
1215 * 334002 : video4linux shouldn't depend on X in configure script
1216 * 336881 : [libvisual] additional support for libvisual-0.4
1217 * 337544 : [xvimagesink] Internal Error when image is too large
1218 * 339520 : [subparse] add " encoding " property
1219 * 340909 : [alsasink] can't enable spdif output
1220 * 341542 : some users have an assertion failed: (GST_VIDEO_SINK_WIDT...
1221 * 341562 : audioconvert doesn't list formats in order of preference
1222 * 341696 : audioconvert crashes if converting from a format with no ...
1223 * 341719 : bisection algorithm in ogg doesn't bisect in some cases
1224 * 341732 : [alsasink] doesn't query supported sample rates
1225 * 341873 : [alsasink] minor memory leak, uses unprotected static var...
1226 * 342143 : [subparse] sami parser needs to escape characters
1227 * 342181 : [alsa] add property probe interface to alsasink and alsasrc
1228 * 342268 : [playbin] add 'subtitle-encoding' property
1229 * 342345 : [riff] Elephant's Dream AVI does not play, JUNK chunk bef...
1230 * 342566 : Building without GTK+ fails
1231 * 343397 : H.264/AAC movie deadlocks with totem in gstreamer code, p...
1232 * 339935 : [adder] dead-locks when adding sink pads in PAUSED state
1234 Changes since 0.10.6:
1236 * typefind improvements
1237 * bug-fixes in textoverlay, audioconvert, videotestsrc,
1238 multifdsink and audio source/sink base classes
1239 * Ice-cast metadata support has moved from gnomevfssrc to the
1240 icydemux element in gst-plugins-good
1241 * audioresample now supports floating point samples
1242 * Adder element fixes.
1243 * Fixes for network playback and audio resampling in playbin
1245 Bugs fixed since 0.10.6:
1247 * 340060 : [adder] handle newsegment events properly
1248 * 340375 : [API 0.11] [patch] typefind to differentiate between mp4 ...
1249 * 339405 : [textoverlay] can't display '\n' character
1250 * 338657 : [patch] adder should send events from src-pad to all sink...
1251 * 338919 : [patch] alsasink should also query witdh capabilities fro...
1252 * 301759 : [audioresample] float audio support (for OSX audio sinks)
1253 * 331901 : [videotestsrc] framerate=0/1 gives assertion error
1254 * 333657 : Replacing icy demuxing in gnomevfssrc
1255 * 336339 : [audioresample] should support width != 16
1256 * 338718 : [patch] [audioconvert] correctly clip float samples > 1.0
1257 * 338778 : [patch] Bad audio with ASX files
1258 * 338991 : [patch] Videoscale doesn't pass on pixel-aspect ratio
1259 * 339574 : [patch] Race condition in multifdsink can lead to spuriou...
1260 * 339786 : [typefinding] wavpack typefinding doesn't always work
1261 * 340369 : [volume element] " volume " property range insufficient
1262 * 340379 : [playbin] doesn't insert audioresample, causes problems w...
1263 * 340392 : Problem with internal-decodebin
1264 * 341160 : [multifdsink] client_status enum has an uninitialized nick
1265 * 341182 : Accessing playbin's streaminfo property from high languag...
1266 * 341432 : [playbin] automatically get icecast metadata requiring ic...
1267 * 341542 : some users have an assertion failed: (GST_VIDEO_SINK_WIDT...
1268 * 341557 : Map GST_TAG_IMAGE < = > ID3v2 APIC tag
1270 API added since 0.10.6:
1272 * client-fd-removed signal added to multifdsink
1273 * stream-info-value-array property added to playbin
1274 * gst_video_calculate_display_ratio() in libgstvideo
1276 Changes since 0.10.5:
1278 * QoS in sinks and transform elements
1279 * Needs GStreamer 0.10.5 for new GstBaseSink::async_playback() vmethod
1280 * added theoraparse element
1282 Bugs fixed since 0.10.5:
1284 * 313136 : [playbin] hang while playing truncated ogg file
1285 * 172848 : [subparse] subtitles with special chars are displayed as ...
1286 * 305279 : [riff] uncompressed AVIs with 24bpp don't work
1287 * 320765 : [ffmpegcolorspace] make win32+msvc compliant, don't use _...
1288 * 323852 : Disable tests/icles on platforms that do not have X
1289 * 325653 : build errors compiling audioresample on win32(vs7)
1290 * 327357 : gst-plugins-base fails to compile with GCC 4.1
1291 * 334620 : [gnomevfssrc] fails to connect to icecast streaming servers
1292 * 334822 : [ffmpegcolorspace] YVU9 support
1293 * 335028 : [typefinding] ID3 v1 tag is not recognized with mp3-in-wa...
1294 * 335365 : inefficient use of GList in gst-plugins-base
1295 * 336190 : [gnomevfssink] should accept non-URI filenames as " location "
1296 * 336194 : [gnomevfssrc] some minor memory leaks
1297 * 336477 : plugins need better/univied descriptions
1298 * 336617 : Unable to recognise MPEG TS stream
1299 * 337548 : Memory leaks in basertpdepayload
1300 * 337945 : [oggdemux] segment stop position ignored
1301 * 338419 : Regression in the handling of files with multiple audio/s...
1302 * 338897 : Videoscale crashes as part of DVD to Ogg transcoding
1303 * 339013 : [videorate] Goes into an infinite loop
1304 * 339047 : [riff] handle H264 fourcc in addition to h264
1305 * 339212 : ISO file typefinding regression
1306 * 330748 : deadlock in base audio sink on playing- > paused state change
1308 Bugs fixed since 0.10.4:
1310 * 334216 : [gnomevfssrc] won't open some media on NFS mounts any longer
1311 * 334226 : typefindfunctions plugin crashes on PPC on registration
1313 Changes since 0.10.3:
1315 * (Experimental) QoS support
1316 * oggmuxer now creates 100% valid streams for Theora, Vorbis and Speex
1317 * documentation updates
1318 * better support for subtitles (seeking)
1320 Bugs fixed since 0.10.3:
1322 * 310202 : [subtitles] < i > < /i > tags and others should be supported i...
1323 * 312439 : XVideo output doesn't work on remote displays (probably r...
1324 * 321271 : audio output is truncated at EOS
1325 * 321650 : Can't decode this ogm file
1326 * 325732 : [oggdemux] problem when seeking to time less than 4s with...
1327 * 325972 : [typefinding] doesn't recognise this mp3
1328 * 326720 : [alsasink] doesn't support more than 2 channels anymore
1329 * 330711 : [ffmpegcolorspace] problems with palettized RGB (fencount...
1330 * 330789 : gstbaseaudiosink causes noise on seeking
1331 * 330888 : Fix build with gcc 2.95 (again)
1332 * 331295 : gnomevfssink doesn't respect umask when creating files
1333 * 331526 : 3GP type detection is too simple
1334 * 331678 : Decodebin is not reusable within a single pipeline (as in...
1335 * 331690 : playbin won't play my last.fm stream
1336 * 331763 : [alsamixer] unmute sets the volume to 100%
1337 * 331765 : [alsamixer] mixer applet slider doesn't want to move from...
1338 * 331903 : [videorate] doesnt handle input caps of framerate=0/1 sanely
1339 * 332778 : [ogmparse] " Already an existing pad " WARNING
1340 * 332964 : random crashes in mp3_type_find
1341 * 333254 : theora encoder does not set IN_CAPS flag properly
1342 * 333352 : [gnomevfssink] reports disk full as generic error
1343 * 333488 : Allow for palette < 256 colours in AVI files
1344 * 333510 : [PATCH] Fix gst_pad_new_from_template (gst_static_pad_tem...
1345 * 333545 : [riff] set depth on wma caps to make asfdemux and pitfdll...
1346 * 333663 : [patch] unref the result of gst_pad_get_parent
1347 * 333900 : [typefind] cannot play a particular mp3 file
1348 * 334112 : variable not initialized
1349 * 334129 : Disable frame dropping for now
1350 * 317038 : use default channel layout if none is specified in multic...
1351 * 319340 : [cdparanoia] uncorrected-error signal never fired
1353 API added since 0.10.3:
1355 * GstTextOverlay::halignment
1356 * GstTextOverlay::valignment
1358 Changes since 0.10.2:
1360 * typefind improvements
1361 * Ogg decoding and encoding fixes
1362 * Improved audio and video sink classes
1363 * Bug and leak fixes
1364 * Improved video scaling
1365 * On-the-fly visualisation switching
1368 Bugs fixed since 0.10.2:
1370 * 330244 : gsttextoverlay.c:895: 'struct _GstCollectData' has no mem...
1371 * 324000 : [playbin] post error or message on unknown input
1372 * 153004 : [typefind] can't identify mp3 file with one single mpeg f...
1373 * 323874 : [playbin] leaks sinks and threads when using gconfaudiosink
1374 * 324626 : ffmpegcolorspace support for fourcc " UYVY "
1375 * 326447 : check that all elements in -base pass queries they can't ...
1376 * 328263 : Fix build with gcc 2.95
1377 * 328279 : [decodebin] timeout issue when pre-rolling
1378 * 329326 : Fix oggmux removing pads from collect pads
1380 Changes since 0.10.1:
1382 * ported gnomevfssink, cdparanoia
1383 * New library and base class: GstCddaBaseSrc
1384 * ported mixerutils.h
1385 * added 'sine-tab' waveform to audiotestsrc
1386 * added float audio to audiorate
1388 Bugs fixed since 0.10.1:
1390 * 324216 : [cdparanoia] missing patches from 0.8
1391 * 324696 : [videotestsrc] does not start counting the time from zero...
1392 * 324900 : Problem compiling gst-plugins-base with Forte
1393 * 325984 : [playbin] cannot handle sources that produce raw audio/video
1394 * 325990 : patch videotestsrc for using glib types
1395 * 326601 : GstRingBuffer crashes with alaw/mulaw caps
1396 * 327114 : [theoradec] should post tags on the bus
1397 * 327216 : vorbisdec segfaults on certain queries
1399 API added since 0.10.1:
1401 * added libgstcddabase
1402 * added mixerutils.h
1404 Changes since 0.10.0:
1406 * Parallel installability with 0.8.x series
1407 * Threadsafe design and API
1408 * removed gst-launch-ext
1410 * Fixes for: subparse, xvimagesink, audioresample, videorate, decodebin
1412 Bugs fixed since 0.10.0:
1414 * 322347 : GstBaseRtpDepayload timestamps are wring
1415 * 323900 : Basertpdepayloader lets NEWSEGMENT events through unfiltered
1416 * 323878 : missing < string.h > inclusion (for memset & FD_ZERO)
1418 API added since 0.10.0:
1420 * GstAlsaMixer::device
1421 * GstAlsaMixer::device-name
1423 Bugs fixed since 0.9.7:
1425 * 319172 : gstreamer-plugins-base-0.9.pc doesn't export linking flags
1426 * 323017 : While(1) loop with sleep(0) in basertpdepayload.c
1428 Changes since 0.9.6:
1430 * Parallel installability with 0.8.x series
1431 * Threadsafe design and API
1432 * ximagesink and xvimagesink updates and interactive test
1434 * rename net to netbuffer library
1435 * rtp element renaming
1436 * stream selector fixes
1438 Bugs fixed since 0.9.6:
1440 * 319618 : [decodebin] some ogg videos don't play
1441 * 320644 : RTP packetizer does't set the packet timestamps correctly
1442 * 322388 : xvimagesink force-aspect-ratio=True always displays squar...
1443 * 322704 : oggdemux typefind list leak
1445 Changes since 0.9.5:
1447 * Parallel installability with 0.8.x series
1448 * Threadsafe design and API
1449 * lots of leak fixes
1450 * flicker-free and rewritten X sinks
1451 * fractional framerates
1452 * removed sinesrc, replaced by audiotestsrc
1454 Bugs fixed since 0.9.5:
1456 * 316442 : playbin should use autoaudiosink/autovideosink by default
1457 * 318353 : [ffmpegcolorspace] forward-port fixes from 0.8 branch
1458 * 320200 : vorbisenc: min-bitrate and max-bitrate are 1/1000 bps rat...
1459 * 321164 : gstringbuffer stops working under load
1460 * 321426 : ximage plugin should be renamed to ximagesink
1461 * 321446 : sinesrc should be dropped in favour of audiotestsrc
1462 * 321451 : GstRtpBuffer: no way to create a sub buffer with only the...
1463 * 321816 : [API] xoverlay API to post prepare-xwindow-id message
1464 * 321894 : vorbisenc doesn't compile
1465 * 322117 : Rename libgsttagedit to libgsttag
1467 Changes since 0.9.4:
1469 * video caps now use a good range for framerate and w/h
1470 * oggdemux/oggmux improvements
1471 * playbin improvements
1473 Bugs fixed since 0.9.4:
1475 * 319110 : [PATCH] oggdemux chain finding is slow
1476 * 320058 : playbin of a jpeg over http does not work
1477 * 320923 : [volume] doesn't build on Solaris
1478 * 321011 : gstbasertpdepayload doesn't send the " new segment " event ...
1480 Changes since 0.9.3:
1482 * New element: audiotestsrc
1483 * typefind improvements
1484 * buffer-frames removed
1486 Changes since 0.9.2:
1490 Bugs fixed since 0.9.2:
1492 * 313251 : ximagesink unused functions
1493 * 315159 : audioconvert lost 24 bit conversions in the rewrite