1 This is GStreamer Base Plug-ins 0.10.25, "Standard disclaimers apply"
5 * Add per-stream volume controls
6 * Theora 1.0 and Y444 and Y42B format support
7 * Improve audio capture timing
8 * GObject introspection support
9 * Improve audio output startup
11 * Use pango-cairo instead of pangoft2
12 * Allow cdda://(device#)?track URI scheme in cddabasesrc
13 * Support interlaced content in videoscale and ffmpegcolorspacee
14 * Many other bug fixes and improvements
16 Bugs fixed since 0.10.24:
18 * 595401 : gobject assertion and null access to volume instance in playbin
19 * 563828 : [decodebin2] Complains about loops in the graph when demuxer output requires another demuxer
20 * 591677 : Easy codec installation is not working
21 * 588523 : smarter sink selection in playbin2
22 * 590146 : adder regressions
23 * 321532 : [cddabasesrc] Support device setting in cdda:// URI
24 * 340887 : add pangocairo textoverlay plugin.
25 * 397419 : [oggdemux] ogm video with subtitles stuck on first frame
26 * 556537 : [PATCH] typefind: more flexible MPEG4 start code recognition
27 * 559049 : gstcheck.c:76:F:general:test_state_changes_* failure: GST_IS_CLOCK(clock) assertion fails
28 * 567660 : [API] need a stream volume interface for sinks that do volume control
29 * 567928 : Make videorate work with a live source
30 * 571610 : [playbin] Scale of volume property is not documented
31 * 583255 : [playbin2] deadlock when disabling visualisations
32 * 586180 : RTSP improvements
33 * 588717 : [oggmux] gst_caps_unref() warning if not linked downstream
34 * 588761 : [videoscale] Needs special support for interlaced content
35 * 588915 : audioresample's output offset counter's initialization could maybe be improved
36 * 589095 : [appsrc] clarify documentation on caps and linkage
37 * 589574 : [typefind] incorrect sdp file detection
38 * 590243 : [videoscale] Claims to support MAX width/height
39 * 590425 : Slaved alsasrc clock with slave-method=re-timestamp not usable for RTP audio
40 * 590856 : [decodebin2] triggers assertion failure on NULL caps
41 * 591207 : totem does display the following subtitle srt file.
42 * 591357 : gst-plugins-base git won't build due to warning in gstrtspconnection.c
43 * 591577 : [playbin2] Incorrect error message string
44 * 591664 : [playbin2] after seeking, srt subtitles don't resync correctly
45 * 591934 : timestamp drift in audioresample
46 * 592544 : Remove regex.h check
47 * 592657 : [appsink] Blocks after entering on pause state
48 * 592864 : deadlocks from recent inputselector/streamselector change
49 * 592884 : [playbin2] g_object_get increases refcount by 2 and therefore leaves memleak
50 * 593035 : gdp doesn't preserve fields of the buffers put into the caps' streamheader
51 * 593284 : basertppayloader takes time in instance init
52 * 594020 : Totem don't play videos from ssh remote host
53 * 594094 : Playback Error playing Midi file
54 * 594136 : [alsasink] Regression from 0.10.23 -- element reuse doesn't work
55 * 594165 : [theoraenc] Implement support for new formats
56 * 594256 : improved slave-skew resynch mechanism
57 * 594258 : missing break in rtcpbuffer
58 * 594275 : Add cast to navigation to fix compiler warning
59 * 594623 : Expose playsink as a fully-fledged element
60 * 594732 : parse error
61 * 594757 : build fails due to warning in gstbasertppayload.c
62 * 594993 : [introspection] pkg-config file madness
63 * 594994 : [streamvolume] Add get_type function to the documentation
64 * 595454 : [cddabasesrc] uri format change breaks rhythmbox
65 * 545807 : [baseaudiosink] audible crack when starting the pipeline
67 API added since 0.10.24:
69 * gst_rtsp_connection_create_from_fd()
70 * gst_rtsp_connection_set_http_mode()
71 * gst_rtsp_watch_write_data()
72 * gst_rtsp_watch_send_message()
73 * GstBaseRTPPayload::perfect-rtptime
74 * GstBaseRTPAudioPayload::gst_base_rtp_audio_payload_flush()
75 * GstVideoSinkClass::show_frame()
76 * GstVideoSink:show-preroll-frame
77 * GST_MIXER_TRACK_READONLY
78 * GST_MIXER_TRACK_WRITEONLY
79 * GstStreamVolume interface
81 Changes since 0.10.23:
83 * Recognise Kate subpicture subtitles
84 * Support progressive download in playbin2
86 * Add buffer-list support in appsink
87 * Add gaussian-noise mode to audiotestsrc
88 * bump cdparanoia req to 0.10.2 and improve caching
89 * Improve audio source base class
90 * Add frame-by-frame stepping and examples
91 * Extend stream-probing in decodebin2
92 * Many RTSP improvements
93 * support for PGS subpictures
95 * Add Y444, v210, v216 formats
96 * implement preset interface in vorbisenc, theoraenc, oggmux
97 * Improve libvisual visualisation timestamp tracking
98 * playbin2 enhancements: custom audiosink, subpictures, cdda
99 * Improvements in textrender
100 * Support raw YUV 4:2:2 and SIREN in RIFF
101 * Add 4:2:2 and 4:4:4 support to theoradec
102 * Many other bug-fixes and improvements
104 Bugs fixed since 0.10.23:
106 * 510417 : [gio] make non-experimental
107 * 513373 : [PATCH] [gstvorbistag] Preserve cover art in Ogg/Vorbis tags
108 * 529300 : [giosink] [PATCH] Allow overwrite
109 * 531035 : [cdparanoia] Should depend on LGPL'd version of the libra...
110 * 567997 : [patch] add allow-pull-scheduling property to audio sinks
111 * 576552 : [subparse] post GST_TAG_SUBTITLE_CODEC tags
112 * 577637 : [playbin2] expose temp-location property
113 * 579692 : mp3_type_find is over-optimistic
114 * 580318 : [tagdemux] drops tag events from upstream
115 * 581460 : [baseaudiosrc] Reusing audio source leads to null timesta...
116 * 581571 : ARGB and alignment added to textrender
117 * 582021 : autogen: libtoolize must be called before aclocal
118 * 582749 : uridecodebin caps property not implemented yet
119 * 582819 : multifdsink: add num-fds property
120 * 583867 : gdpdepay + identity cause failed assertions
121 * 584020 : [playbin2] inadvertently resets configured audio/video sinks
122 * 584686 : [playbin2] Need {audio,video,text}-tags-changed signals
123 * 585197 : [subparse] fails to detect subrip subtitles with fewer th...
124 * 585758 : Remove deprecated GTK+ symbols
125 * 585970 : gst_audioringbuffer_get_type is not thread safe
126 * 585994 : gst-rtsp-message doesn't support " Timestamp " filed
127 * 586331 : [cdparanoia] expose cd cache size parameter
128 * 586356 : [playbin2] use private copy of input-selector as long as ...
129 * 586519 : white Gaussian noise would be useful in audiotestsrc
130 * 587080 : rtsp fails to compile - doesn't see some ws2tcpip functions
131 * 587278 : Support for GstBufferList in appsink
132 * 587676 : Call tzset() before localtime_r(), in e.g. gst-plugins-ba...
133 * 587695 : Patches to add stream-status messages audio elements
134 * 587896 : " No stream given yet " error from giostreamsrc
135 * 587980 : gstchannelmix.c: protect debug code with GST_DISABLE_GST_...
136 * 588078 : [playbin2] Fails to go to READY again after an error
137 * 588205 : Pipeline with giostreamsrc will not enter playing state
138 * 588550 : build failure in git, missing gstinterfaces-0.10
139 * 588551 : queue2: download buffering fixes
140 * 588724 : [vorbisdec] empty encoder string causes GStreamer
141 * 588746 : [audiotestsrc] Make sure tags are properly serialized in ...
142 * 588747 : [adder] Serialize incoming in-band events (tags) in the d...
143 * 588748 : [adder] Check dataflow consistency in unit tests
144 * 589075 : [playbin2] changing volume doesn't work after stream rest...
145 * 589581 : typefinder: recognise more Kate subtitle categories
146 * 589622 : Cannot use both playbin and input-selector
147 * 589663 : gstreamer asserts in gstaudiofilter
148 * 589797 : alsasrc does not set GstAlsaSrc- > handle to NULL after snd...
149 * 590470 : [typefinding] certain flac-in-ogg files not detected any ...
150 * 536313 : [cdda] Remove sha1 copy once we depend on glib-2.16
151 * 579642 : [oggdemux] handle broken ogg/vorbis files better
152 * 582528 : playbin2 Audio CD playback broken since
153 * 583318 : Assertion from within playbin2
154 * 585079 : undefined references to gst_adapter_* functions in schro
155 * 585708 : [adder] Wrong handling of flushing seeks
156 * 588218 : Siren in .wav support
157 * 586920 : rtsp: needs < netinet/in.h > on FreeBSD
159 API added since 0.10.23:
161 * GstNetAddress::gst_netaddress_to_string()
162 * Add gst_rtsp_watch_queue_data()
163 * playbin2: Add {audio,video,text}-tags-changed signals
164 * Add gst_color_balance_get_balance_type()
165 * Add gst_mixer_get_mixer_type()
167 Changes since 0.10.22:
169 * New navigation API to support DVD playback
170 * playbin2 improvements
171 * RTSP extensions to allow extra headers and options
172 * Replace audioresampler with speexresample based code
173 * Support interlacing flags in the gstvideo library
174 * Support new RIFF formats
175 * Improve typefinding
176 * Support more frame formats in videoscale
177 * Many other bug-fixes and improvements
179 Bugs fixed since 0.10.22:
181 * 577637 : [playbin2] expose temp-location property
182 * 580120 : [playbin2] unit test fails
183 * 478512 : [alsamixer] volume control slider not working
184 * 574962 : rhythmbox crash in flac_type_find
185 * 564139 : Documentation of TCP plugins
186 * 577436 : xvimagesink should use xcontext- > depth and not count bits...
187 * 350311 : [playbin2] support for subpicture subtitles
188 * 378094 : Enable pango elements to handle UYVY
189 * 543591 : Gnonlin can not play theora streams
190 * 553295 : [riff] fuzzed AVI file causes segfault
191 * 565105 : Gstreamer does not change from READY back to PAUSED in sa...
192 * 565777 : [riff] unrecognised video fourcc 0x10000002 for mpeg2 in avi
193 * 566661 : [typefind] Fall back to file extension using uri query
194 * 567255 : [riff] doesn't detect codec_id 0x706d as AAC (amongst other)
195 * 567636 : [pbutils] Missing plugins code shouldn't ask for the same...
196 * 567740 : bogus warning in decodebin2?
197 * 568482 : linking problems in gst-plugins-base
198 * 569655 : [ffmpegcolorspace] Add UYVY422 to GRAY8 conversion function
199 * 570142 : Documentation is broken for uridecodebin
200 * 570356 : aac typefinder failure
201 * 570768 : [ximagesink] wrong mouse pointer position if output windo...
202 * 570832 : Add flags to enhance mixer interfaces
203 * 571009 : [tagdemux] WMA file with id3v2 tag causes assertion to fail
204 * 571147 : [ffmpegcolorspace/videotestsrc] Add support for packed/pl...
205 * 572577 : [playbin2] deadlock on shutdown
206 * 572872 : [ffmpegcolorspace] Add YVYU colorspace
207 * 572993 : [subparse] broken libregex dependency on Windows
208 * 573165 : Generate additional export files for gstreamer app plugin
209 * 573528 : Wrong format modifier in gstgiobasesink.c
210 * 573529 : In gstrtspconnection.c some functions are called with wro...
211 * 574293 : [decodebin2] deadlock on shutdown
212 * 574319 : Missing HAVE_PROCESS_H in win32/common/config.h
213 * 574447 : gstadder.c: line 904: error C2036: 'gpointer' : unknown size
214 * 574939 : [typefinding] flac typefinder mis-typefinds PDFs as flac ...
215 * 575550 : srt subtitle file keeps playbin2 from playing
216 * 575638 : kissfft copyright
217 * 575649 : [oggdemux] duration query in time format returns true wit...
218 * 576019 : On Windows queue2 can't write files longer than 2-4 GiB, ...
219 * 576142 : [vorbisenc] Non-header output buffers have NULL caps
220 * 576180 : [playbin2] Uses unref'd audiosink volume if using gconfau...
221 * 576586 : [alsamixer] gnome-sound-properties freeze
222 * 577054 : [videoscale] Not valgrind clean
223 * 577709 : Review new navigation API
224 * 577827 : [appsink] Have appsink new_buffer-callback return GstFlow...
225 * 578583 : [PATCH] multifdsink doesn't handle sync-method=latest-key...
226 * 578656 : Implement upstream GstForceKeyUnit events in theoraenc
227 * 579129 : pkgconfig: appsrc/appsink can not be linked to uninstalled
228 * 579130 : app: expose trivial type macros
229 * 579192 : gst_rtcp_packet_get_type should not assert on packet content
230 * 579203 : baseaudiosink: unparenting the ringbuffer in NULL causes ...
231 * 579267 : [rtspconnection] g_async_queue_new_full() is GLib-2.16 AP...
232 * 579463 : [cddabasesrc] [cdparanoiasrc] no longer emits discid
233 * 579668 : audioresample fails to build with --disable-gst-debug
234 * 579734 : [playbin] raw_decoding_mode seems to be set unconditionally
235 * 579912 : [decodebin2] multiqueue is too small in time (interleave ...
236 * 580470 : [audioresample] causes pipelines to go out of sync and be...
237 * 580952 : [audioresample] bad quality/pops compared to plughw
238 * 581727 : [playbin2] make playsink go to PAUSED async
239 * 569682 : playbin2 leaks request pad from input selector
240 * 580020 : [vorbisenc] causes buffers to be out of segment if new se...
241 * 562794 : rtspsrc fails to create a socket on Win32 sometimes.
242 * 567396 : playbin2: DECODE_BIN_LOCK occasionally called twice withi...
243 * 567982 : " queued_bytes " field isn't updated while flushing the que...
244 * 571299 : [appsink] Handoff callback API
245 * 574443 : rtsp win32 - forgotten variable
246 * 574516 : [typefind] add typefinder for photoshop .psd files
247 * 574964 : gst_app_src_end_of_stream(), mutex on error return
248 * 575256 : rtspsrc fails to resolve hostnames
249 * 575588 : decodebin2 deadlock
250 * 576187 : [playbin2] Stalls video sink when disabling subtitles in ...
251 * 576188 : [playbin2] Reusing a playbin2 instance with visualization...
252 * 576190 : [playbin2] Deadlock when reusing playbin2 after an error
253 * 577288 : " Internal playbin error " when seeking to the end of files
254 * 577610 : RTCP feedback messages support in GstRTCPPacket
255 * 577794 : [playbin2] leaks elements set through properties
256 * 578118 : [multifdsink] add option to not resend the streamheader w...
257 * 578506 : Pipeline with alsasrc and alsasink cannot change state ba...
258 * 578942 : Missing RTSP headers related to Windows Media extension.
259 * 580271 : videorate: fails to clear discont flag on duplicated buffers
260 * 580649 : uridecodebin: bug on documentation published in website
262 API added since 0.10.22:
264 * GstRTSP::gst_rtsp_options_as_text()
265 * GstRTSPMessage::gst_rtsp_message_take_header()
266 * GstRTSPRange::gst_rtsp_range_to_string()
267 * New Navigation interface commands, queries and messages
268 * gst_rtsp_channel_new()
269 * gst_rtsp_channel_unref()
270 * gst_rtsp_channel_attach()
271 * gst_rtsp_channel_queue_message()
272 * gst_rtsp_connection_accept()
273 * GstAppSink::gst_app_sink_set_callbacks()
274 * GST_VIDEO_FORMAT_YVYU,GST_VIDEO_BUFFER_TFF,GST_VIDEO_BUFFER_RFF,GST_VIDEO_BUFFER_ONEFIELD
275 * GST_MIXER_FLAG_HAS_WHITELIST,GST_MIXER_FLAG_GROUPING,GST_MIXER_TRACK_NO_RECORD,GST_MIXER_TRACK_NO_MUTE,GST_MIXER_TRACK_WHITELIST
276 * GstAppSrc::emit-signals
277 * GstAppSrc::gst_app_src_set_emit_signals()
278 * GstAppSrc::gst_app_src_get_emit_signals()
279 * GstAppSrc::gst_app_src_set_callbacks()
280 * RTSP::gst_rtsp_connection_get_url()
281 * GstRTSPLowerTrans::GST_RTSP_LOWER_TRANS_HTTP
282 * RTSP:gst_rtsp_connection_set_tunneled()
283 * RTSP:gst_rtsp_connection_is_tunneled()
284 * RTSP::gst_rtsp_connection_set_ip()
285 * RTSP::gst_rtsp_connection_get_tunnelid()
286 * RTSP::gst_rtsp_connection_do_tunnel()
287 * RTSP::gst_rtsp_watch_reset()
291 1) Please note that decodebin2 and playbin2 API included in this release is
292 still considered unstable and WILL change in future releases. At this stage,
293 only developers or early adopters should consider using decodebin2 or playbin2
294 API embodied in their signals and properties.
296 Changes since 0.10.21:
298 * Require gettext 0.17
299 * Replace audioresample with speexresample from -bad
300 * Support new formats in RIFF: uncompressed RGB, WMA lossless, VP6
301 * Move libgstapp and elements from -bad
302 * Support color-key setting and probing for Xv properties
303 * Improve typefinding for various formats
304 * Extend audio sinks for pull-mode operation
305 * Support for more subtitle formats
306 * More development on decode2bin and playbin2
308 * Many bug fixes and improvements
310 Bugs fixed since 0.10.21:
312 * 562163 : theoraenc likely ignoring segments
313 * 562258 : rtspsrc element takes long time to error out if the addre...
314 * 561789 : [volume] deadlocks with a controller attached
315 * 554533 : [xvimagesink] allow setting colorkey if possible
316 * 567511 : colorkey in xvimagesink gets reset when element is reused
317 * 116051 : libresample doesn't handle > factor of 2 rate conversion
318 * 346218 : [audioresample] doesn't do anti aliasing
319 * 385061 : [audioresample?] investigate high CPU usage
320 * 456788 : [subparse] can't handle UTF-16 charset encoded subtitle.
321 * 525807 : [vorbisenc] vorbisenc has problems with a gnlsource that ...
322 * 546955 : gstoggmux EOS handling issue
323 * 549417 : [audioresample] unit test fails on 64bit linux
324 * 549510 : audioresample doesn't negotiate ideal caps
325 * 552237 : UTF-16 srt confuses gstreamer, misdetected as mp3
326 * 552559 : Implementation of SLAVE_SKEW in baseaudiosrc
327 * 552569 : audioresample producing strange sized buffers
328 * 552801 : audioconvert can overflow with big audio buffers
329 * 554879 : Add ability to specify format for date/time display in Gs...
330 * 555257 : Doesn't display srt subtitles saved with BOM
331 * 555319 : add FFV1 fourcc to riff-media
332 * 555607 : subrip subtitles typefind too strict
333 * 555699 : [PATCH] theoradec: prefer container's pixel aspect ratio ...
334 * 556025 : build failure in tests/icles
335 * 556066 : Last byte of FLAC image buffer chopped off
336 * 557365 : subparse check fails
337 * 558124 : [PLUGIN-MOVE] Move speexresample as audioresample2 to -base
338 * 559111 : ALSA sink hangs on USB audio device unplug while playing
339 * 559478 : does not play windows media streams correctly
340 * 559567 : `gst_base_audio_sink_sync_latency' should call `gst_base_...
341 * 561436 : videorate element add image/jpeg to caps template
342 * 561734 : playbin2 additions
343 * 561780 : Playbin2 should work without volume too
344 * 561924 : oggdemux hangs when given corrupt input via non-seekable ...
345 * 562270 : build without gdk fails
346 * 563143 : ximagesink/xvimagesink : _alloc_buffer returns non-clean ...
347 * 563174 : Implement gst_rtcp_packet_remove
348 * 563508 : [rgvolume] Unit test fails with passthrough assertions
349 * 563718 : Theora check out of date
350 * 563904 : GNOME Goal: Clean up GLib and GTK+ includes
352 API added since 0.10.21:
354 * clockoverlay::time-format
355 * GstRingBuffer:gst_ring_buffer_activate()
356 * GstRingBuffer:gst_ring_buffer_is_active()
357 * GstRingBuffer:gst_ring_buffer_convert()
358 * Add GST_TYPE_BASE_AUDIO_(SRC|SINK)_SLAVE_METHOD to the public API
359 * gst_netaddress_get_address_bytes()
360 * gst_netaddress_set_address_bytes()
362 Changes since 0.10.20:
364 * Continue playbin2 development
365 * Ogg improvements - CELT support, skeleton fixes
366 * DVD subpicture support
367 * Improved audio dithering random number generator
368 * xvimagesink/ximagesink fixes
369 * Vorbis encoding and decoding fixes
370 * Recognise Kate subtitle streams
371 * Many bug-fixes and enhancements
373 Bugs fixed since 0.10.20:
375 * 537380 : [gnomevfssrc] Doesn't handle short reads properly
376 * 538656 : xvimagesink support for autofill/colorkey property
377 * 540334 : Build fails without X in tests/examples/seek
378 * 528299 : Multiple GstMixerTracks with the same label cause problem...
379 * 530068 : Ogg Streams with Skeleton and Granulepos > 0 do not work(...
380 * 537009 : playbin2 silly typo breaks signals
381 * 537045 : decodebin2 sometimes emits 'drained' multiple times
382 * 537599 : [oggdemux] skeleton streams not skipped in ogg
383 * 537889 : [xvimagesink] colorbalance is bad
384 * 538232 : vorbisenc/vorbisdec don't work with a live source
385 * 538663 : gdppay memleak in gst_gdp_pay_reset
386 * 540215 : decodebin does not insert a queue for raw data type
387 * 540351 : [avidemux] Doesn't know about Duck DK4 ADPCM
388 * 540497 : ffmpegcolorspace is returning wrong size
389 * 541358 : cross mingw32 gcc: getaddrinfo is not in ws2_32.dll befor...
390 * 544306 : rtspsrc debug=1 segfaults with some libc
391 * 548898 : GStreamer-CRITICAL errors on seeking beyond stream borders
392 * 548913 : vorbisenc being picky about rounding errors in timestamps
393 * 549062 : Video devices aren't updated on subsequent probing.
394 * 549814 : [typefind] add application/pdf typefinder
395 * 550582 : [oggdemux] KATE streams not recognised
396 * 550638 : [typefind] Recognize some jpeg2k file types
397 * 550656 : recognize TrueSpeech in wavparse
398 * 550729 : gst-plugins-base won't compile with " -pedantic " option
399 * 552960 : tagdemux asserts and aborts on truncated files
400 * 553244 : theoraparse doesn't work at all (throws criticals and ass...
402 API added since 0.10.20:
404 * Add "index" property to GstMixerTrack to differantiate between
405 multiple mixer tracks with the same label.
407 Changes since 0.10.19:
410 * Support digest auth for RTSP
411 * Additional documentation
412 * Support DSCP QoS in multifdsink
413 * Add NV12/NV21 video buffer layouts
414 * Video scaling now bilinear by default
415 * Support more than 8 channels in audio conversions
416 * Channel mapping fixes for audioconvert
417 * Improve tmplayer and sami subtitle support
418 * Support 1x1 pixel buffers for videoscale
419 * Typefinding improvements for MPEG2, musepack
420 * Ogg/Dirac mapping updated in oggmux
421 * Fixes in ogg demuxing
422 * audiosink synchronisation and slaving fixes
423 * Support muting of the audio in playbin by selecting -1 as the audio stream
424 * Work done on playbin2 and uridecodebin
425 * Improvements in the experimental GIO plugin
427 * Handle GAP buffers in some places
428 * Various other leak and bug-fixes
430 Bugs fixed since 0.10.20:
432 * 526794 : [giosrc] totem doesn't work with some gvfs backends
433 * 510417 : [PLUGIN-MOVE] Move gio to gst-plugins-base
434 * 509125 : crash in CD Player: - playing CD - lowering/...
435 * 517813 : [audioconvert] make gap aware
436 * 302798 : [playbin] add mute property
437 * 342294 : Setting playbin property current-audio=-1 also stops the ...
438 * 398033 : [audioconvert] support more than 8 channels
439 * 419351 : [avi/a52dec] AV synchronization problems
440 * 467911 : [subparse] sami parser update
441 * 469933 : multifdsink IPv6 and diffserv TOS/TC markup
442 * 506659 : [textoverlay] rendering error when using non-standard widths
443 * 512333 : [gstvorbistag] Retrieve Ogg/Vorbis cover art as image met...
444 * 512382 : [playbin] race condition when pausing/playing multiple in...
445 * 518037 : pbutils-enumtypes.c is not included in win32/vs6/libgstpb...
446 * 521761 : gstaudioclock frozen the clock value until reaches latest...
447 * 522401 : gdpdepay doesn't validate payload CRCs
448 * 523993 : playbin2 blocks after a while when listening to a radio s...
449 * 524724 : [PATCH] [baseaudiosrc] buffer-time and latency-time do no...
450 * 525665 : Crash on Ogg/Vorbis with chain=NULL
451 * 525915 : [streamheader] Unit test fails with " gst_adapter_peek: as...
452 * 526173 : [typefinding] fails to detect mpeg video stream whereas m...
453 * 529018 : gst_ogm_parse_stream_header creates fraction value with w...
454 * 529500 : [videotestsrc] support for NV12 and NV21
455 * 529546 : [Playbin] Memory leak in streaminfo handling
456 * 530068 : Ogg Streams with Skeleton and Granulepos > 0 do not work(...
457 * 530531 : [typefinding] bad read in mpeg_video_stream_type_find
458 * 530719 : gst_video_calculate_display_ratio fails when playing Ogg ...
459 * 530962 : [subparse] parses only every second line of TMPlayer subt...
460 * 532454 : [NV12/NV21] videotestsrc and ffmpegcolorspace don't play ...
461 * 533087 : GstRTSPTransport kept opaque in docs
462 * 533817 : [audioconvert] Can't use default 7 channel layout / only ...
463 * 534071 : Gdppay memleak
464 * 534331 : race in decodebin when changing states while the internal...
465 * 535356 : vorbisdec doesn't support 8 channels
466 * 536475 : gdppay memleak and possible crash
467 * 536521 : Refcounting errors in playbin
468 * 536874 : Build failure on windows
469 * 532166 : [ffmpegcolorspace] support NV12 format
470 * 533617 : [audioconvert] Produces silence when converting 1/2 chann...
471 * 536848 : [giosrc] Doesn't handle short reads properly
472 * 536849 : [giosrc] Very slow doing any playback
473 * 518082 : [alsamixer] playback volumes overwritten by capture volum...
474 * 435633 : [PATCH] videorate not (fully) segment aware; causes frame...
475 * 532364 : tcpclientsrc broken in 0.10.19
476 * 533075 : gst_rtp_buffer_compare_seqnum doesn't do what it says
477 * 533265 : [cddabasesrc] Sound Juicer cut a sector when ripping a track
479 API additions since 0.10.20:
481 * decodebin2::sink-caps property
482 * giosrc::file property
483 * giosink::file property
484 * gst_base_audio_src_set_slave_method()
485 * gst_base_audio_src_get_slave_method()
486 * GstAudioClock::gst_audio_clock_reset()
487 * GstBaseAudioSrc:actual-buffer-time property
488 * GstBaseAudioSrc:actual-latency-time property
489 * gst_audio_check_channel_positions()
490 * add gst_tag_image_data_to_image_buffer()
491 * add gst_tag_list_add_id3_image()
492 * add GST_TAG_IMAGE_TYPE_NONE enum value
494 Changes since 0.10.18:
496 * Handle EAGAIN when polling sockets in rtspconnection
498 Changes since 0.10.17:
500 * Experimental GIO plugin
501 * Continued playbin2 development
503 * Better network element support on Windows
504 * Various other bug-fixes and improvements
506 Bugs fixed since 0.10.17:
508 * 509637 : [API] [basertpaudiopayload] add _set_samplebits_options()
509 * 510229 : [gnomevfssrc] HTTPS support
510 * 511478 : [rtpbuffer] add gst_rtp_buffer_set_extension_data function
511 * 511810 : [RTSP] Uses MT-unsafe gmtime() function
512 * 512899 : [alsa] gstalsasink.c:527: warning: 'snd_pcm_sw_params_set...
513 * 513167 : Fix compiler warning due to disabled signals in mixertrac...
514 * 514307 : [playbin] warning in nautilus, volume element can't be cr...
515 * 514623 : Ogg Theora video slow
516 * 514937 : Correct initialization of hints in is_multicast_address()
517 * 515654 : xvimagesink doesn't build with --disable-xshm
518 * 516246 : [alsasink] handle negative delay from snd_pcm_delay
519 * 517420 : typefind: add h264 elementary stream discovery
520 * 517991 : problems with configure file depending on GCC compiler
521 * 518039 : libgstrtsp MSVC 6.0 compile error
522 * 518162 : [subparse] handle italic text starting with " / " with Micr...
523 * 518940 : [playbin2] make _get_*_tags() match vfuncs prototype in c...
524 * 519906 : [API] add GstMixerOptions::get_values vfunc
525 * 519916 : [API] add mixer-changed and options-list-changed messages
526 * 520523 : [API] Unreviewed changes to ringbuffer API
527 * 521743 : libgstnetbuffer.def exports not up to date
528 * 522625 : [video] gst_video_format_parse_caps() broken for RGBA for...
529 * 523054 : gstbasesrc crashes when called from typefind helpers
530 * 511825 : [RTSP] compiler warning on FreeBSD
531 * 520300 : [alsasrc] provide-clock=false messes up buffer durations
533 API added since 0.10.17:
535 * GstRTPBuffer:gst_rtp_buffer_set_extension_data()
536 * add GST_VIDEO_FORMAT_Y41B and GST_VIDEO_FORMAT_Y42B.
537 * add GstMixerOptions::get_values vfunc (#519906)
538 * add gst_mixer_options_list_changed(), gst_mixer_mixer_changed() and
539 gst_mixer_message_parse_options_list_changed(). Fixes #519916.
540 * gst_base_rtp_audio_payload_set_samplebits_options()
541 * GstNetBuffer::gst_netaddress_equal
543 Changes since 0.10.16:
545 * Work-around ABI breakage due to unfortunate use of the
546 GST_DISABLE_DEPRECATED macro
547 * Export 2 missing functions needed for bindings in the win32 build
548 * Initialise the GstRingBuffer GType from a thread-safe context
550 Bugs fixed since 0.10.16:
552 * 511825 : [RTSP] compiler warning on FreeBSD
553 * 513018 : crash in Volume Control: I typed my password at t...
554 * 512334 : g_critical() when using GstAudioFilter & GST_DEBUG
556 Changes since 0.10.15:
558 * Handle newer Theora granule-pos semantics
559 * Introducing first alpha version playbin2 - the upcoming successor to
561 * Fixes in playbin handling of stream-switching
562 * New API for uniform handling of raw-video format buffers.
563 * Improvements for RTSP/RTP handling
564 * RIFF lib additions for VC-1 and AVC1 fourccs
565 * Many other bug-fixes and improvements
567 Bugs fixed since 0.10.15:
569 * 506132 : Review of changes in video/video.h
570 * 320984 : [oggdemux] cannot handle multiple chains
571 * 373011 : [playbin] throws error when switching off subtitles
572 * 436756 : Intermittent crashes in Pidgin in audioclock g_type_class...
573 * 462740 : [streamselector] patch to improve default stream selection
574 * 486840 : [alsamixer] use _all variants when setting the mixer
575 * 497964 : theoraenc test fails
576 * 498228 : gst-plugins-base-0.10.15 does not compile on FreeBSD (Gen...
577 * 499697 : Provide better pkg-config files
578 * 502497 : [subparse] SubRip subtitles starting from 0 not recognised
579 * 503440 : The control sockets used by gstrtspconnection.c are never...
580 * 503930 : [cdda] warning: 'eos' may be used uninitialized in this f...
581 * 506928 : [alsamixer] add " PCM " as master fall back for cards that ...
582 * 508138 : [decodebin] does not error out if pad activation fails
583 * 509762 : missing file in win32/MANIFEST
584 * 511274 : gst_rtp_buffer_get_extension_data is returning FALSE when...
585 * 496731 : [PATCH] xvimagesink leaks memory if initialization fails
586 * 496761 : [PATCH] RTSP message leaks memory when uninitialized
587 * 500763 : SIGSEGV while playing ogg audio file
589 API additions since 0.10.15:
591 * New GstVideoFormat API and helper functions in libgstvideo
592 * gst_base_audio_sink_set_provide_clock()
593 * gst_base_audio_sink_get_provide_clock()
594 * gst_base_audio_sink_set_slave_method()
595 * gst_base_audio_sink_get_slave_method()
596 * gst_base_audio_src_set_provide_clock()
597 * gst_base_audio_src_get_provide_clock()
599 Changes since 0.10.14:
601 * RTP/RTSP/RTCP/SDP support improved
602 * New FFT support library libgstfft, based on Kiss FFT
603 * New formats supported in volume and audiotestsrc
604 * Fixes in audiorate and videorate
605 * Audio capture fixes
606 * Playbin and decodebin fixes
607 * New tagdemux base class for ID3/APE style tag readers
608 * Fix a nasty crash in the X sinks on shutdown
610 * Add support for multichannel WAV files.
611 * Preserve channel layout information when up/down-mixing.
612 * Many bug-fixes and improvements
614 Bugs fixed since 0.10.14:
616 * 475395 : decodebin2 leaks request-pads
617 * 475451 : [decodebin2] leaks ghostpad
618 * 378770 : [xvimagesink] race condition in event thread?
619 * 407282 : [decodebin2] autoplug-sort signal has GList ** parameter
620 * 430677 : [audioconvert] does not preserve channel positions when f...
621 * 442654 : [volume] controller bypassed by default
622 * 445529 : [volume] support for 24/32-bit audio/x-raw-int
623 * 446766 : return code for gst_base_rtp_payload_audio_handle_event()
624 * 451970 : Subparse requires HTML parser
625 * 453650 : [audiobasesrc] two alsasrcs do not work in one pipeline
626 * 459334 : [textoverlay] expose pango line alignment property
627 * 459585 : [basertpdepayload] api without namespace
628 * 460422 : [audiotestsrc] Add support for float and double output
629 * 462805 : [alsa] compilation fails with gcc 4.2
630 * 462979 : Add 'silent' property to GstTimeOverlay
631 * 463215 : [audioconvert] compile errors
632 * 464320 : [PATCH] gst-plugins-base-0.14 does not build for win32
633 * 464666 : [playbin] QT trailer hangs in preroll with decodebin2
634 * 464690 : Add connection-speed property to uridecodebin element
635 * 465015 : [playbin] Not removed probes causes deadlocks in streamin...
636 * 465028 : some warnings with mingw
637 * 467667 : GST_FRAMES_TO_CLOCK_TIME() and GST_CLOCK_TIME_TO_FRAMES()...
638 * 468129 : [basertpaudiopayload] event handler returns the wrong value
639 * 468619 : New library gstfft: FFT library for integer and float typ...
640 * 470456 : [API] add gst_missing_*_installer_detail_new()
641 * 470766 : [ssaparse] line breaks in SSA subtitle parser
642 * 471067 : Make the SDP code useable for generating SDP descriptions
643 * 471194 : [rtpbuffer] RTP headers are wrong for win32
644 * 473097 : [baseaudiosink] gstreamer-properties hangs when testing s...
645 * 474384 : gstrtsp-enumtypes.c and .h needed for win32
646 * 474880 : [xvimagesink] [ximagesink] leaking buffer caps reference
647 * 475731 : rtspconnection is able to read incomplete messages
648 * 483620 : All Rtp buffers are discarded -- gst_rtp_buffer_get_payl...
649 * 484989 : memleak, not unrefed caps for gstbasertppayload.c
650 * 489010 : Please change default channel order for WAVE_EXT-less .wa...
651 * 491722 : [playbin] regression: crash with external subtitles
652 * 492098 : [GstFFT] Broken scaling
653 * 492114 : Build issues on Windows/MSVC
654 * 492306 : compilation errors with MinGW
655 * 492813 : Missing symbols in libgstrtp.def
656 * 493986 : Build issues on Windows (missing symbols)
657 * 494346 : pre-release vs6 patch
658 * 496548 : Including malloc.h breaks macos build
659 * 496724 : DSW file references non-existent DSP files
660 * 464079 : audiotestsrc doesn't respond to conversion queries properly
661 * 442065 : floatcast.h includes config.h and might break other apps
662 * 466717 : gst_event_new_new_segment_full:assertion `start < = stop' ...
663 * 485753 : Decodebin2 deadlocks when nulling pipeline during typefind
664 * 464028 : Move connection-speed from playbin to playbasebin
666 API added since 0.10.14:
668 * GstTagDemux base class for simple tag demuxers
669 * GstBaseAudioSrc::provide-clock property
670 * gst_rtcp_ntp_to_unix()
671 * gst_rtcp_unix_to_ntp()
672 * gst_rtp_buffer_get_header_len()
673 * gst_rtp_buffer_get_extension_data()
674 * gst_rtp_buffer_compare_seqnum()
675 * gst_rtp_buffer_ext_timestamp()
676 * gst_rtcp_packet_sdes_copy_entry()
677 * gst_install_plugins_supported()
678 * gst_missing_*_installer_detail_new() convenience API
679 * gst_rtsp_connection_poll()
680 * GstTextOverlay::line-alignment property
682 Changes since 0.10.13:
684 * Audio dither and noise-shaping when reducing bit-depth
685 * RTSP and SDP helper libraries added
686 * Experimental buffering element "queue2" now supports pull-mode
687 and file-based buffering.
688 * Support for more 32-bit video pixel layouts
689 * Various fixes and improvements
691 Bugs fixed since 0.10.13:
693 * 380625 : [x*imagesink] add 'handle-expose' property
694 * 385527 : oggmux sometimes gets DELTA flag on output wrong near start
695 * 402076 : videoscale 4-tap method broken for downscaling
696 * 437169 : [xvimagesink] add property to disable Xv double-buffering
697 * 441264 : queue2 support to do buffering on a file
698 * 442553 : [v4lsrc] doesn't output segments in GST_FORMAT_TIME
699 * 442557 : [videorate] doesn't handle latency queries
700 * 442944 : Audiotestsrc can overflow on seeks
701 * 444523 : [queue2] Pull mode support
702 * 444630 : Compilation error with fsseko (from gstqueue2.c) -- unabl...
703 * 445505 : [queue2] It does not work in pull mode with oggdemux
704 * 446551 : [queue2] Buffering is not working properly if it is set t...
705 * 446572 : [queue2] Division by zero
706 * 446972 : warning when compiling gstoggdemux.c
707 * 449156 : Regression in CVS for decodebin2
708 * 450875 : Missing files in po/POTFILES.in
709 * 451707 : [tag] UTF-8 in ID3v1 tag not correctly decoded
710 * 451908 : [ffmpegcolorspace] regression: doesn't accept GST_VIDEO_C...
711 * 454264 : Playbin fails to " play " image url after a movie url
712 * 456656 : [API] Addition of audio buffer clipping function to gstaudio
713 * 460978 : gst_audio_buffer_clip outputs warnings
714 * 152864 : [PATCH] GstAlsaMixer doesn't support signals
715 * 360246 : [audioconvert] Optionally apply dithering
716 * 394061 : Add support for Subviewer subtitles
717 * 420326 : Base payloader class has wrong property types and ranges
718 * 451145 : [vorbisdec] errors out on 0-sized packets
719 * 459204 : [PATCH] [playbin] gst_play_base_bin_get_streaminfo_value_...
721 API added since 0.10.13:
723 * RTSP and SDP libraries added
724 * gst_rtsp_base64_decode_ip
725 * Add buffer clipping function gst_audio_buffer_clip for raw audio
726 buffers. Fixes #456656.
727 * gst_mixer_get_mixer_flags
728 * gst_mixer_message_parse_mute_toggled
729 * gst_mixer_message_parse_record_toggled
730 * gst_mixer_message_parse_volume_changed
731 * gst_mixer_message_parse_option_changed
732 * GstMixerMessageType
735 Changes since 0.10.12:
736 * Many fixes and improvements
737 * RTP and RTCP support improved
739 Bugs fixed since 0.10.12:
741 * 339838 : [audioconvert] support floats with non-native endianness
742 * 393975 : closing x/xvimagesink window crashes gst-launch
743 * 405072 : [API] add gst_tag_freeform_string_to_utf8()
744 * 413799 : [subparse] add support for MPL2 format
745 * 414645 : GstMixerTrack should make untranslated label available
746 * 420079 : [audioconvert] Uses biased rounding which results in dist...
747 * 420578 : [subparse] add more colour map in sami parser
748 * 421834 : videorate breaks on dimension changes
749 * 423051 : Vorbis tags of type double use locale-dependent formatting
750 * 423055 : Verify ReplayGain vorbistag processing in libs/tag testsuite
751 * 425455 : Decodebin2 leaks pads
752 * 426250 : GstPlayBaseBin leaks streaminfo objects
753 * 428187 : Rtp base depayloader class doesn't send new_segment after...
754 * 431672 : gst_base_rtp_audio_payload_push() should take object of i...
755 * 432362 : [ximagesink] doesn't build if XShm is not available
756 * 432755 : [videorate] leaks buffer if flow != OK
757 * 432984 : [baseaudiosrc] misleading warning message when dropping s...
758 * 433888 : [theoradec] does not generate a perfect stream
759 * 436562 : Theoradec doesn't work well with gnonlin
760 * 438840 : [theoradec] does not compile with old version of libtheora
761 * 440997 : [gstriff] Doesn't handle width!=depth files with audio/x-...
762 * 441295 : audioconvert doesn't build on VS6
763 * 442024 : regression in playbin buffering
764 * 350299 : [playbin] " Internal data flow error " opening movie with s...
765 * 410039 : totem crashed with SIGSEGV in new_decoded_pad_full()
766 * 340842 : do latency calculation for live sources
767 * 341078 : RB does not play beyond initially downloaded podcast file
768 * 414496 : [id3demux, id3v2mux] Add support for GST_TAG_MUSICBRAINZ_...
770 API additions since 0.10.12:
772 * add gst_tag_freeform_string_to_utf8()
773 * GstRTPBuffer::gst_rtp_buffer_default_clock_rate()
774 * GstBaseAudioSink::slave-method property
775 * add "min-ptime" property to RTP base audio payloader
776 * gst_base_rtp_audio_payload_push()
777 * gst_base_rtp_audio_payload_get_adapter()
778 * GstMixerTrack::untranslated-label property
780 Changes since 0.10.11:
782 * New API for on-demand plugin installation
783 * Xv thread-safety and configuration enhancements
784 * decodebin2 improvements
785 * Support more raw audio format conversions
786 * Improvements in Ogg support
787 * AudioFilter base class ported to 0.10
788 * Fixes for subtitles
789 * Latency/live-playback support for Alsa
790 * Lots of bug fixes and improvements
792 Bugs fixed since 0.10.11:
794 * 398721 : No video in .ogm files with decodebin2
795 * 339837 : [audioconvert] support for 64-bit float audio
796 * 341524 : [decodebin] can't handle decoders with always src pads wi...
797 * 352069 : Add de.po German translation
798 * 363379 : [oggmux] doesn't detect EOS on all sinkpads
799 * 378436 : [oggdemux] rhythmbox crash on fast clicking on rating in ...
800 * 380342 : Totem does not play mp3 files when lyrics are present
801 * 383195 : [cddabasesrc,basertpaudiopayload] compile errors with gcc...
802 * 383198 : totem crashed to gst_xvimagesink_update_colorbalance
803 * 384008 : [xvimagesink] accesses - > xwindow outside locks
804 * 384060 : gst_xoverlay_set_xwindow_id() causing lockups with x(v)im...
805 * 387138 : x input events processing in sinks with xoverlay interfac...
806 * 390063 : Documentation typo
807 * 390076 : add xv adaptor and port properties in xvimagesink element.
808 * 391365 : [oggdemux] internal stream error on OggFlac
809 * 392070 : [vorbis] GST_TAG_LOCATION not mapped
810 * 392393 : [API] add libgstbaseutils library for missing plugins mes...
811 * 396042 : mpeg4 video typefinder loops endlessly on quicktime redirect
812 * 396835 : audioconvert/audioresample combination causing buffer of ...
813 * 397673 : [patch] XIOError caught in x[v]imagesink.c
814 * 397810 : [typefinding] .vob file: could not determine type of stream
815 * 398110 : [theoraenc] GLib failed to allocate 3080991032 bytes on g...
816 * 399340 : Crash in the oggdemux plugin when trying to play a specia...
817 * 401029 : [playbin] rapidly changing visualisation freezes
818 * 401072 : Move libgimme-codec helper functions to GStreamer
819 * 402505 : visualisations don't work for some samplerates
820 * 407811 : decodebin2 hang on HD clip
821 * 409683 : Crash with Decodebin2
822 * 410396 : not reading " DATE " tags from Flac files
823 * 410963 : Fails to build with -z defs
824 * 357503 : [suparse] wrong timing with microdvd subtitles
825 * 393310 : [pango] localtime_r does not exist in MinGW
826 * 397207 : Test failure w/ HP-UX 11.11 & native compiler
827 * 399948 : [textoverlay] leaks upstream events if textpad unlinked
828 * 403963 : GstAudioFilter base class broken
829 * 404512 : [videoscale] floating point exception on 1x1 video
830 * 405020 : [alsa] probing the device-name doesn't seem to work corre...
831 * 408278 : [videorate] memory leak
832 * 410772 : Crash copying a GstNetBuffer
833 * 401118 : [visual] error if width not a multiple of 4
834 * 405451 : [alsasink] deadlocks when disconnecting USB Sounddevice
836 API additions since 0.10.11:
839 * GST_VIDEO_SINK_CAST()
840 * gst_pb_utils_add_codec_description_to_tag_list()
841 * gst_pb_utils_get_codec_description()
842 * gst_pb_utils_get_source_description()
843 * gst_pb_utils_get_sink_description()
844 * gst_pb_utils_get_decoder_description()
845 * gst_pb_utils_get_encoder_description()
846 * gst_pb_utils_get_element_description()
847 * gst_pb_utils_init()
848 * gst_install_plugins_context_new()
849 * gst_install_plugins_context_set_xid()
850 * gst_install_plugins_context_free()
851 * gst_install_plugins_async()
852 * gst_install_plugins_sync()
853 * gst_install_plugins_return_get_name()
854 * gst_install_plugins_installation_in_progress()
855 * gst_missing_uri_source_message_new()
856 * gst_missing_uri_sink_message_new
857 * gst_missing_element_message_new
858 * gst_missing_decoder_message_new
859 * gst_missing_encoder_message_new
860 * gst_missing_plugin_message_get_installer_detail
861 * gst_missing_plugin_message_get_description
862 * gst_is_missing_plugin_message
864 Bugs fixed since 0.10.10:
866 * 360552 : [riff] [avi] extracts non-UTF8 metadata
867 * 365501 : [x/xvimagesink] race condition when creating first image ...
868 * 339366 : [playbin] hangs if suburi file type cannot be determined
869 * 355914 : libvisual causes xvimagesink: assertion `GST_CAPS_REFCOU...
870 * 363118 : gst_riff_create_video_caps() should also store variant in...
871 * 363607 : xvimagesink xwindow_draw_border() slowness
872 * 336301 : [playbin] can't handle RTSP source
873 * 337026 : oggmux doesn't set EOS properly
874 * 337031 : vorbisdec outputs too much data
875 * 340049 : New BaseRTPAudioPayloader class to -base
876 * 348264 : Theora encoding, Ogg muxing don't handle discontinuities
877 * 354773 : xvimage assumes that XV_COLORKEY can be set in RGB888 format
878 * 355917 : libvisual plugin is broken
879 * 355935 : multifdsink doesn't allow setting maximums (soft, hard) i...
880 * 357038 : [ffmpegcolorspace] RGBA handling broken
881 * 357215 : [playbin] buffering notification not quite right yet
882 * 357289 : [riff] riff parser can't detect aac audio stream
883 * 357404 : [playbin] Linking can fail silently
884 * 357531 : [subparse] problem if markup is not closed
885 * 357577 : [playbin] regression: buffering still images broken
886 * 357591 : Avoid compiler warning with uclibc and -Werror
887 * 357613 : XvStopVideo in xvimagesink
888 * 357800 : [libvisual] doesn't pass audio data to libvisual 0.4.0 co...
889 * 359580 : tcpserversink and dataprotocol assert for multipart streams
890 * 361095 : Fixes compiling with forte: warning clean up (part 3)
891 * 361456 : [basertppayload] Memory leak
892 * 361634 : sink- > ringbuffer NULL in BaseAudioSink's setcaps()
893 * 361984 : [subparse] doesn't accept .srt file that doesn't start wi...
894 * 366334 : [PATCH] Windows vs8 fixes
895 * 368273 : Using the remove signal on multifdsink is not threadsafe
896 * 368310 : include file gstbasertpaudiopayload.h not included for r...
897 * 369482 : [typefind] MPEG system streams get recognized as mp3 files
898 * 370092 : [PATCH] Decodebin v2 : Implementation
899 * 377183 : regression: no eos when playing ogg vorbis files
900 * 381219 : bad debugging code left in audiorate
901 * 382223 : [decodebin] more delayed linking
902 * 382269 : Typefind detects mpeg video clip as audio/mpeg
903 * 335635 : Add an Ogg/Vorbis retagging element
904 * 341681 : [textoverlay] flickering with continuously timestamped text
905 * 342228 : [alsa] Recognize " Front " as a Master channel
906 * 357330 : [subparse] some sami parser minor but enhanced patch
907 * 357532 : [gsttag] vorbistag doesn't handle dates that include time...
908 * 359237 : [typefinding] doesn't recognize XML files shorter than 25...
909 * 362845 : [subparse] add support for tmplayer format
910 * 357977 : [videorate] new segment start is not respected
911 * 364812 : [PATCH] oggmux release pad does not remove pad
912 * 364856 : pngenc stride problems
913 * 372507 : Mac build fixes
915 API added since 0.10.10:
917 * playbin::queue-min-threshold property.
918 * GstVideoOrientation interface
919 * gst_base_rtp_depayload_push_ts
920 * gst_base_rtp_depayload_push
921 * Add dropped_buffers to multifdsink's get-stats GValueArray
922 * gst_ring_buffer_commit_full
924 Changes since 0.10.9:
926 * New elements: gdppay, gdpdepay
928 Bugs fixed since 0.10.9:
930 * 343787 : The adder cannot handle when multiple elements tries to l...
931 * 336075 : ALSA emu10k1 mixer tracks are wrongly classified as playb...
932 * 349105 : crash with playbin and resizing screen
933 * 342494 : [v4l] Query " device-name " even if device is not open
934 * 342680 : [adder] seeking with multiple ogg files fails to work
935 * 345188 : [alsa] can't handle more than 8 channels
936 * 347091 : converting vorbis comments to GstTagLists is lossy
937 * 348157 : Changed " Change Device " menu behaviour in gnome-volume-co...
938 * 348916 : [typefind] add multipart/x-mixed-replace typefinder
939 * 350157 : [riff] riff parser can't detect dts audio stream
940 * 350655 : [oggdemux] should process seeking queries
941 * 350900 : [adder] should not clamp floating point values
942 * 351426 : API: add gst_tag_parse_extended_comment
943 * 351502 : g_value_set_string leaks
944 * 351742 : [vorbisenc] discontinuity detection too sensitive, might ...
945 * 353658 : [videotestsrc] doesn't round strides correctly for YVYU
946 * 354594 : multifdsink doesn't work reliably with sync-method = 'nex...
947 * 351790 : [ogmparse] crash parsing video stream on x86-64
948 * 140139 : [avidemux] can't play broken avi with ogg (not vorbis) au...
949 * 347783 : [PLUGIN-MOVE] GDP elements should be moved
950 * 347918 : Internal data flow error in udpsrc
951 * 349656 : jitterbuffer in GstBaseRtp fails to handle rtp seqnum rol...
952 * 350784 : element alsamixer doesn't respect asoundrc
953 * 351308 : [netbuffer] build fails with gkt-doc critical warnings
954 * 353234 : audiorate preserves DISCONT on buffers
955 * 353912 : Add cmml caps to oggmux
957 API added since 0.10.9:
959 * gst_rtp_buffer_get_payload_subbuffer()
960 * gst_tag_parse_extended_comment()
961 * GstPlayBin::connection-speed
962 * GstTheoraParse::synchronization-points
963 * GST_AUDIO_CHANNEL_POSITION_NONE
965 Changes since 0.10.8:
967 * Parallel installability with 0.8.x series
968 * Threadsafe design and API
970 * Support for images in tags
971 * Playback improvements
972 * Gnomevfssrc now supports burn:// uris
973 * Videoscale now supports more RGBA formats
974 * Multifdsink improvements
975 * Testsuite can now generate coverage information
977 Bugs fixed since 0.10.8:
979 * 347296 : Problems with clocks on alsasrc hangs the application
980 * 347295 : [vorbisdec] Pushes before being initialized
981 * 329798 : [playbin] doesn't always give correct error message for m...
982 * 342085 : [alsasink] doesn't set buffer-time correctly
983 * 342789 : [audioresample] doesn't clear state when stopped, causing...
984 * 343303 : [subparse] workaround for bad entities in sami parser
985 * 343385 : [gnomevfs] add support for burn:// URIs
986 * 343500 : [riff] gst_riff_parse_strf_vids() can't parse extra data.
987 * 343699 : oggmux leaks
988 * 344503 : [subparse] parse font face property in sami parser.
989 * 345131 : [PATCH] videoscale support for 32-bit RGB-formats
990 * 345206 : [textoverlay] crash with non-UTF8 input
991 * 345225 : [theoradec] Clipping for exact seeking
992 * 345641 : [API] [libgsttag] add enums for image tag type
993 * 345879 : [riff] won't play a .wmv file with WMVA video stream
994 * 346581 : [typefinding] recognise text/html
995 * 347221 : [audioconvert] channel remapping does not work right
996 * 347304 : Massive leaks with xvimagesink
997 * 346527 : alsasrc get_range does not respect requested size
999 Changes since 0.10.7:
1001 * alsasink probing fixes
1002 * xvimagesink error reporting fixes
1005 * vorbis multichannel fixes
1006 * multifdsink streamheader fixes
1008 Bugs fixed since 0.10.7:
1010 * 169936 : [subparse] support for SAMI subtitles
1011 * 315312 : Gstreamer Xv uses RGB instead of YUV.
1012 * 334002 : video4linux shouldn't depend on X in configure script
1013 * 336881 : [libvisual] additional support for libvisual-0.4
1014 * 337544 : [xvimagesink] Internal Error when image is too large
1015 * 339520 : [subparse] add " encoding " property
1016 * 340909 : [alsasink] can't enable spdif output
1017 * 341542 : some users have an assertion failed: (GST_VIDEO_SINK_WIDT...
1018 * 341562 : audioconvert doesn't list formats in order of preference
1019 * 341696 : audioconvert crashes if converting from a format with no ...
1020 * 341719 : bisection algorithm in ogg doesn't bisect in some cases
1021 * 341732 : [alsasink] doesn't query supported sample rates
1022 * 341873 : [alsasink] minor memory leak, uses unprotected static var...
1023 * 342143 : [subparse] sami parser needs to escape characters
1024 * 342181 : [alsa] add property probe interface to alsasink and alsasrc
1025 * 342268 : [playbin] add 'subtitle-encoding' property
1026 * 342345 : [riff] Elephant's Dream AVI does not play, JUNK chunk bef...
1027 * 342566 : Building without GTK+ fails
1028 * 343397 : H.264/AAC movie deadlocks with totem in gstreamer code, p...
1029 * 339935 : [adder] dead-locks when adding sink pads in PAUSED state
1031 Changes since 0.10.6:
1033 * typefind improvements
1034 * bug-fixes in textoverlay, audioconvert, videotestsrc,
1035 multifdsink and audio source/sink base classes
1036 * Ice-cast metadata support has moved from gnomevfssrc to the
1037 icydemux element in gst-plugins-good
1038 * audioresample now supports floating point samples
1039 * Adder element fixes.
1040 * Fixes for network playback and audio resampling in playbin
1042 Bugs fixed since 0.10.6:
1044 * 340060 : [adder] handle newsegment events properly
1045 * 340375 : [API 0.11] [patch] typefind to differentiate between mp4 ...
1046 * 339405 : [textoverlay] can't display '\n' character
1047 * 338657 : [patch] adder should send events from src-pad to all sink...
1048 * 338919 : [patch] alsasink should also query witdh capabilities fro...
1049 * 301759 : [audioresample] float audio support (for OSX audio sinks)
1050 * 331901 : [videotestsrc] framerate=0/1 gives assertion error
1051 * 333657 : Replacing icy demuxing in gnomevfssrc
1052 * 336339 : [audioresample] should support width != 16
1053 * 338718 : [patch] [audioconvert] correctly clip float samples > 1.0
1054 * 338778 : [patch] Bad audio with ASX files
1055 * 338991 : [patch] Videoscale doesn't pass on pixel-aspect ratio
1056 * 339574 : [patch] Race condition in multifdsink can lead to spuriou...
1057 * 339786 : [typefinding] wavpack typefinding doesn't always work
1058 * 340369 : [volume element] " volume " property range insufficient
1059 * 340379 : [playbin] doesn't insert audioresample, causes problems w...
1060 * 340392 : Problem with internal-decodebin
1061 * 341160 : [multifdsink] client_status enum has an uninitialized nick
1062 * 341182 : Accessing playbin's streaminfo property from high languag...
1063 * 341432 : [playbin] automatically get icecast metadata requiring ic...
1064 * 341542 : some users have an assertion failed: (GST_VIDEO_SINK_WIDT...
1065 * 341557 : Map GST_TAG_IMAGE < = > ID3v2 APIC tag
1067 API added since 0.10.6:
1069 * client-fd-removed signal added to multifdsink
1070 * stream-info-value-array property added to playbin
1071 * gst_video_calculate_display_ratio() in libgstvideo
1073 Changes since 0.10.5:
1075 * QoS in sinks and transform elements
1076 * Needs GStreamer 0.10.5 for new GstBaseSink::async_playback() vmethod
1077 * added theoraparse element
1079 Bugs fixed since 0.10.5:
1081 * 313136 : [playbin] hang while playing truncated ogg file
1082 * 172848 : [subparse] subtitles with special chars are displayed as ...
1083 * 305279 : [riff] uncompressed AVIs with 24bpp don't work
1084 * 320765 : [ffmpegcolorspace] make win32+msvc compliant, don't use _...
1085 * 323852 : Disable tests/icles on platforms that do not have X
1086 * 325653 : build errors compiling audioresample on win32(vs7)
1087 * 327357 : gst-plugins-base fails to compile with GCC 4.1
1088 * 334620 : [gnomevfssrc] fails to connect to icecast streaming servers
1089 * 334822 : [ffmpegcolorspace] YVU9 support
1090 * 335028 : [typefinding] ID3 v1 tag is not recognized with mp3-in-wa...
1091 * 335365 : inefficient use of GList in gst-plugins-base
1092 * 336190 : [gnomevfssink] should accept non-URI filenames as " location "
1093 * 336194 : [gnomevfssrc] some minor memory leaks
1094 * 336477 : plugins need better/univied descriptions
1095 * 336617 : Unable to recognise MPEG TS stream
1096 * 337548 : Memory leaks in basertpdepayload
1097 * 337945 : [oggdemux] segment stop position ignored
1098 * 338419 : Regression in the handling of files with multiple audio/s...
1099 * 338897 : Videoscale crashes as part of DVD to Ogg transcoding
1100 * 339013 : [videorate] Goes into an infinite loop
1101 * 339047 : [riff] handle H264 fourcc in addition to h264
1102 * 339212 : ISO file typefinding regression
1103 * 330748 : deadlock in base audio sink on playing- > paused state change
1105 Bugs fixed since 0.10.4:
1107 * 334216 : [gnomevfssrc] won't open some media on NFS mounts any longer
1108 * 334226 : typefindfunctions plugin crashes on PPC on registration
1110 Changes since 0.10.3:
1112 * (Experimental) QoS support
1113 * oggmuxer now creates 100% valid streams for Theora, Vorbis and Speex
1114 * documentation updates
1115 * better support for subtitles (seeking)
1117 Bugs fixed since 0.10.3:
1119 * 310202 : [subtitles] < i > < /i > tags and others should be supported i...
1120 * 312439 : XVideo output doesn't work on remote displays (probably r...
1121 * 321271 : audio output is truncated at EOS
1122 * 321650 : Can't decode this ogm file
1123 * 325732 : [oggdemux] problem when seeking to time less than 4s with...
1124 * 325972 : [typefinding] doesn't recognise this mp3
1125 * 326720 : [alsasink] doesn't support more than 2 channels anymore
1126 * 330711 : [ffmpegcolorspace] problems with palettized RGB (fencount...
1127 * 330789 : gstbaseaudiosink causes noise on seeking
1128 * 330888 : Fix build with gcc 2.95 (again)
1129 * 331295 : gnomevfssink doesn't respect umask when creating files
1130 * 331526 : 3GP type detection is too simple
1131 * 331678 : Decodebin is not reusable within a single pipeline (as in...
1132 * 331690 : playbin won't play my last.fm stream
1133 * 331763 : [alsamixer] unmute sets the volume to 100%
1134 * 331765 : [alsamixer] mixer applet slider doesn't want to move from...
1135 * 331903 : [videorate] doesnt handle input caps of framerate=0/1 sanely
1136 * 332778 : [ogmparse] " Already an existing pad " WARNING
1137 * 332964 : random crashes in mp3_type_find
1138 * 333254 : theora encoder does not set IN_CAPS flag properly
1139 * 333352 : [gnomevfssink] reports disk full as generic error
1140 * 333488 : Allow for palette < 256 colours in AVI files
1141 * 333510 : [PATCH] Fix gst_pad_new_from_template (gst_static_pad_tem...
1142 * 333545 : [riff] set depth on wma caps to make asfdemux and pitfdll...
1143 * 333663 : [patch] unref the result of gst_pad_get_parent
1144 * 333900 : [typefind] cannot play a particular mp3 file
1145 * 334112 : variable not initialized
1146 * 334129 : Disable frame dropping for now
1147 * 317038 : use default channel layout if none is specified in multic...
1148 * 319340 : [cdparanoia] uncorrected-error signal never fired
1150 API added since 0.10.3:
1152 * GstTextOverlay::halignment
1153 * GstTextOverlay::valignment
1155 Changes since 0.10.2:
1157 * typefind improvements
1158 * Ogg decoding and encoding fixes
1159 * Improved audio and video sink classes
1160 * Bug and leak fixes
1161 * Improved video scaling
1162 * On-the-fly visualisation switching
1165 Bugs fixed since 0.10.2:
1167 * 330244 : gsttextoverlay.c:895: 'struct _GstCollectData' has no mem...
1168 * 324000 : [playbin] post error or message on unknown input
1169 * 153004 : [typefind] can't identify mp3 file with one single mpeg f...
1170 * 323874 : [playbin] leaks sinks and threads when using gconfaudiosink
1171 * 324626 : ffmpegcolorspace support for fourcc " UYVY "
1172 * 326447 : check that all elements in -base pass queries they can't ...
1173 * 328263 : Fix build with gcc 2.95
1174 * 328279 : [decodebin] timeout issue when pre-rolling
1175 * 329326 : Fix oggmux removing pads from collect pads
1177 Changes since 0.10.1:
1179 * ported gnomevfssink, cdparanoia
1180 * New library and base class: GstCddaBaseSrc
1181 * ported mixerutils.h
1182 * added 'sine-tab' waveform to audiotestsrc
1183 * added float audio to audiorate
1185 Bugs fixed since 0.10.1:
1187 * 324216 : [cdparanoia] missing patches from 0.8
1188 * 324696 : [videotestsrc] does not start counting the time from zero...
1189 * 324900 : Problem compiling gst-plugins-base with Forte
1190 * 325984 : [playbin] cannot handle sources that produce raw audio/video
1191 * 325990 : patch videotestsrc for using glib types
1192 * 326601 : GstRingBuffer crashes with alaw/mulaw caps
1193 * 327114 : [theoradec] should post tags on the bus
1194 * 327216 : vorbisdec segfaults on certain queries
1196 API added since 0.10.1:
1198 * added libgstcddabase
1199 * added mixerutils.h
1201 Changes since 0.10.0:
1203 * Parallel installability with 0.8.x series
1204 * Threadsafe design and API
1205 * removed gst-launch-ext
1207 * Fixes for: subparse, xvimagesink, audioresample, videorate, decodebin
1209 Bugs fixed since 0.10.0:
1211 * 322347 : GstBaseRtpDepayload timestamps are wring
1212 * 323900 : Basertpdepayloader lets NEWSEGMENT events through unfiltered
1213 * 323878 : missing < string.h > inclusion (for memset & FD_ZERO)
1215 API added since 0.10.0:
1217 * GstAlsaMixer::device
1218 * GstAlsaMixer::device-name
1220 Bugs fixed since 0.9.7:
1222 * 319172 : gstreamer-plugins-base-0.9.pc doesn't export linking flags
1223 * 323017 : While(1) loop with sleep(0) in basertpdepayload.c
1225 Changes since 0.9.6:
1227 * Parallel installability with 0.8.x series
1228 * Threadsafe design and API
1229 * ximagesink and xvimagesink updates and interactive test
1231 * rename net to netbuffer library
1232 * rtp element renaming
1233 * stream selector fixes
1235 Bugs fixed since 0.9.6:
1237 * 319618 : [decodebin] some ogg videos don't play
1238 * 320644 : RTP packetizer does't set the packet timestamps correctly
1239 * 322388 : xvimagesink force-aspect-ratio=True always displays squar...
1240 * 322704 : oggdemux typefind list leak
1242 Changes since 0.9.5:
1244 * Parallel installability with 0.8.x series
1245 * Threadsafe design and API
1246 * lots of leak fixes
1247 * flicker-free and rewritten X sinks
1248 * fractional framerates
1249 * removed sinesrc, replaced by audiotestsrc
1251 Bugs fixed since 0.9.5:
1253 * 316442 : playbin should use autoaudiosink/autovideosink by default
1254 * 318353 : [ffmpegcolorspace] forward-port fixes from 0.8 branch
1255 * 320200 : vorbisenc: min-bitrate and max-bitrate are 1/1000 bps rat...
1256 * 321164 : gstringbuffer stops working under load
1257 * 321426 : ximage plugin should be renamed to ximagesink
1258 * 321446 : sinesrc should be dropped in favour of audiotestsrc
1259 * 321451 : GstRtpBuffer: no way to create a sub buffer with only the...
1260 * 321816 : [API] xoverlay API to post prepare-xwindow-id message
1261 * 321894 : vorbisenc doesn't compile
1262 * 322117 : Rename libgsttagedit to libgsttag
1264 Changes since 0.9.4:
1266 * video caps now use a good range for framerate and w/h
1267 * oggdemux/oggmux improvements
1268 * playbin improvements
1270 Bugs fixed since 0.9.4:
1272 * 319110 : [PATCH] oggdemux chain finding is slow
1273 * 320058 : playbin of a jpeg over http does not work
1274 * 320923 : [volume] doesn't build on Solaris
1275 * 321011 : gstbasertpdepayload doesn't send the " new segment " event ...
1277 Changes since 0.9.3:
1279 * New element: audiotestsrc
1280 * typefind improvements
1281 * buffer-frames removed
1283 Changes since 0.9.2:
1287 Bugs fixed since 0.9.2:
1289 * 313251 : ximagesink unused functions
1290 * 315159 : audioconvert lost 24 bit conversions in the rewrite
projects / platform / upstream / gstreamer.git / blob