1 This is GStreamer Base Plug-ins 0.10.24, "Counting the days"
5 * Recognise Kate subpicture subtitles
6 * Support progressive download in playbin2
8 * Add buffer-list support in appsink
9 * Add gaussian-noise mode to audiotestsrc
10 * bump cdparanoia req to 0.10.2 and improve caching
11 * Improve audio source base class
12 * Add frame-by-frame stepping and examples
13 * Extend stream-probing in decodebin2
14 * Many RTSP improvements
15 * support for PGS subpictures
17 * Add Y444, v210, v216 formats
18 * implement preset interface in vorbisenc, theoraenc, oggmux
19 * Improve libvisual visualisation timestamp tracking
20 * playbin2 enhancements: custom audiosink, subpictures, cdda
21 * Improvements in textrender
22 * Support raw YUV 4:2:2 and SIREN in RIFF
23 * Add 4:2:2 and 4:4:4 support to theoradec
24 * Many other bug-fixes and improvements
26 Bugs fixed since 0.10.23:
28 * 510417 : [gio] make non-experimental
29 * 513373 : [PATCH] [gstvorbistag] Preserve cover art in Ogg/Vorbis tags
30 * 529300 : [giosink] [PATCH] Allow overwrite
31 * 531035 : [cdparanoia] Should depend on LGPL'd version of the libra...
32 * 567997 : [patch] add allow-pull-scheduling property to audio sinks
33 * 576552 : [subparse] post GST_TAG_SUBTITLE_CODEC tags
34 * 577637 : [playbin2] expose temp-location property
35 * 579692 : mp3_type_find is over-optimistic
36 * 580318 : [tagdemux] drops tag events from upstream
37 * 581460 : [baseaudiosrc] Reusing audio source leads to null timesta...
38 * 581571 : ARGB and alignment added to textrender
39 * 582021 : autogen: libtoolize must be called before aclocal
40 * 582749 : uridecodebin caps property not implemented yet
41 * 582819 : multifdsink: add num-fds property
42 * 583867 : gdpdepay + identity cause failed assertions
43 * 584020 : [playbin2] inadvertently resets configured audio/video sinks
44 * 584686 : [playbin2] Need {audio,video,text}-tags-changed signals
45 * 585197 : [subparse] fails to detect subrip subtitles with fewer th...
46 * 585758 : Remove deprecated GTK+ symbols
47 * 585970 : gst_audioringbuffer_get_type is not thread safe
48 * 585994 : gst-rtsp-message doesn't support " Timestamp " filed
49 * 586331 : [cdparanoia] expose cd cache size parameter
50 * 586356 : [playbin2] use private copy of input-selector as long as ...
51 * 586519 : white Gaussian noise would be useful in audiotestsrc
52 * 587080 : rtsp fails to compile - doesn't see some ws2tcpip functions
53 * 587278 : Support for GstBufferList in appsink
54 * 587676 : Call tzset() before localtime_r(), in e.g. gst-plugins-ba...
55 * 587695 : Patches to add stream-status messages audio elements
56 * 587896 : " No stream given yet " error from giostreamsrc
57 * 587980 : gstchannelmix.c: protect debug code with GST_DISABLE_GST_...
58 * 588078 : [playbin2] Fails to go to READY again after an error
59 * 588205 : Pipeline with giostreamsrc will not enter playing state
60 * 588550 : build failure in git, missing gstinterfaces-0.10
61 * 588551 : queue2: download buffering fixes
62 * 588724 : [vorbisdec] empty encoder string causes GStreamer
63 * 588746 : [audiotestsrc] Make sure tags are properly serialized in ...
64 * 588747 : [adder] Serialize incoming in-band events (tags) in the d...
65 * 588748 : [adder] Check dataflow consistency in unit tests
66 * 589075 : [playbin2] changing volume doesn't work after stream rest...
67 * 589581 : typefinder: recognise more Kate subtitle categories
68 * 589622 : Cannot use both playbin and input-selector
69 * 589663 : gstreamer asserts in gstaudiofilter
70 * 589797 : alsasrc does not set GstAlsaSrc- > handle to NULL after snd...
71 * 590470 : [typefinding] certain flac-in-ogg files not detected any ...
72 * 536313 : [cdda] Remove sha1 copy once we depend on glib-2.16
73 * 579642 : [oggdemux] handle broken ogg/vorbis files better
74 * 582528 : playbin2 Audio CD playback broken since
75 * 583318 : Assertion from within playbin2
76 * 585079 : undefined references to gst_adapter_* functions in schro
77 * 585708 : [adder] Wrong handling of flushing seeks
78 * 588218 : Siren in .wav support
79 * 586920 : rtsp: needs < netinet/in.h > on FreeBSD
81 API added since 0.10.23:
83 * GstNetAddress::gst_netaddress_to_string()
84 * Add gst_rtsp_watch_queue_data()
85 * playbin2: Add {audio,video,text}-tags-changed signals
86 * Add gst_color_balance_get_balance_type()
87 * Add gst_mixer_get_mixer_type()
89 Changes since 0.10.22:
91 * New navigation API to support DVD playback
92 * playbin2 improvements
93 * RTSP extensions to allow extra headers and options
94 * Replace audioresampler with speexresample based code
95 * Support interlacing flags in the gstvideo library
96 * Support new RIFF formats
98 * Support more frame formats in videoscale
99 * Many other bug-fixes and improvements
101 Bugs fixed since 0.10.22:
103 * 577637 : [playbin2] expose temp-location property
104 * 580120 : [playbin2] unit test fails
105 * 478512 : [alsamixer] volume control slider not working
106 * 574962 : rhythmbox crash in flac_type_find
107 * 564139 : Documentation of TCP plugins
108 * 577436 : xvimagesink should use xcontext- > depth and not count bits...
109 * 350311 : [playbin2] support for subpicture subtitles
110 * 378094 : Enable pango elements to handle UYVY
111 * 543591 : Gnonlin can not play theora streams
112 * 553295 : [riff] fuzzed AVI file causes segfault
113 * 565105 : Gstreamer does not change from READY back to PAUSED in sa...
114 * 565777 : [riff] unrecognised video fourcc 0x10000002 for mpeg2 in avi
115 * 566661 : [typefind] Fall back to file extension using uri query
116 * 567255 : [riff] doesn't detect codec_id 0x706d as AAC (amongst other)
117 * 567636 : [pbutils] Missing plugins code shouldn't ask for the same...
118 * 567740 : bogus warning in decodebin2?
119 * 568482 : linking problems in gst-plugins-base
120 * 569655 : [ffmpegcolorspace] Add UYVY422 to GRAY8 conversion function
121 * 570142 : Documentation is broken for uridecodebin
122 * 570356 : aac typefinder failure
123 * 570768 : [ximagesink] wrong mouse pointer position if output windo...
124 * 570832 : Add flags to enhance mixer interfaces
125 * 571009 : [tagdemux] WMA file with id3v2 tag causes assertion to fail
126 * 571147 : [ffmpegcolorspace/videotestsrc] Add support for packed/pl...
127 * 572577 : [playbin2] deadlock on shutdown
128 * 572872 : [ffmpegcolorspace] Add YVYU colorspace
129 * 572993 : [subparse] broken libregex dependency on Windows
130 * 573165 : Generate additional export files for gstreamer app plugin
131 * 573528 : Wrong format modifier in gstgiobasesink.c
132 * 573529 : In gstrtspconnection.c some functions are called with wro...
133 * 574293 : [decodebin2] deadlock on shutdown
134 * 574319 : Missing HAVE_PROCESS_H in win32/common/config.h
135 * 574447 : gstadder.c: line 904: error C2036: 'gpointer' : unknown size
136 * 574939 : [typefinding] flac typefinder mis-typefinds PDFs as flac ...
137 * 575550 : srt subtitle file keeps playbin2 from playing
138 * 575638 : kissfft copyright
139 * 575649 : [oggdemux] duration query in time format returns true wit...
140 * 576019 : On Windows queue2 can't write files longer than 2-4 GiB, ...
141 * 576142 : [vorbisenc] Non-header output buffers have NULL caps
142 * 576180 : [playbin2] Uses unref'd audiosink volume if using gconfau...
143 * 576586 : [alsamixer] gnome-sound-properties freeze
144 * 577054 : [videoscale] Not valgrind clean
145 * 577709 : Review new navigation API
146 * 577827 : [appsink] Have appsink new_buffer-callback return GstFlow...
147 * 578583 : [PATCH] multifdsink doesn't handle sync-method=latest-key...
148 * 578656 : Implement upstream GstForceKeyUnit events in theoraenc
149 * 579129 : pkgconfig: appsrc/appsink can not be linked to uninstalled
150 * 579130 : app: expose trivial type macros
151 * 579192 : gst_rtcp_packet_get_type should not assert on packet content
152 * 579203 : baseaudiosink: unparenting the ringbuffer in NULL causes ...
153 * 579267 : [rtspconnection] g_async_queue_new_full() is GLib-2.16 AP...
154 * 579463 : [cddabasesrc] [cdparanoiasrc] no longer emits discid
155 * 579668 : audioresample fails to build with --disable-gst-debug
156 * 579734 : [playbin] raw_decoding_mode seems to be set unconditionally
157 * 579912 : [decodebin2] multiqueue is too small in time (interleave ...
158 * 580470 : [audioresample] causes pipelines to go out of sync and be...
159 * 580952 : [audioresample] bad quality/pops compared to plughw
160 * 581727 : [playbin2] make playsink go to PAUSED async
161 * 569682 : playbin2 leaks request pad from input selector
162 * 580020 : [vorbisenc] causes buffers to be out of segment if new se...
163 * 562794 : rtspsrc fails to create a socket on Win32 sometimes.
164 * 567396 : playbin2: DECODE_BIN_LOCK occasionally called twice withi...
165 * 567982 : " queued_bytes " field isn't updated while flushing the que...
166 * 571299 : [appsink] Handoff callback API
167 * 574443 : rtsp win32 - forgotten variable
168 * 574516 : [typefind] add typefinder for photoshop .psd files
169 * 574964 : gst_app_src_end_of_stream(), mutex on error return
170 * 575256 : rtspsrc fails to resolve hostnames
171 * 575588 : decodebin2 deadlock
172 * 576187 : [playbin2] Stalls video sink when disabling subtitles in ...
173 * 576188 : [playbin2] Reusing a playbin2 instance with visualization...
174 * 576190 : [playbin2] Deadlock when reusing playbin2 after an error
175 * 577288 : " Internal playbin error " when seeking to the end of files
176 * 577610 : RTCP feedback messages support in GstRTCPPacket
177 * 577794 : [playbin2] leaks elements set through properties
178 * 578118 : [multifdsink] add option to not resend the streamheader w...
179 * 578506 : Pipeline with alsasrc and alsasink cannot change state ba...
180 * 578942 : Missing RTSP headers related to Windows Media extension.
181 * 580271 : videorate: fails to clear discont flag on duplicated buffers
182 * 580649 : uridecodebin: bug on documentation published in website
184 API added since 0.10.22:
186 * GstRTSP::gst_rtsp_options_as_text()
187 * GstRTSPMessage::gst_rtsp_message_take_header()
188 * GstRTSPRange::gst_rtsp_range_to_string()
189 * New Navigation interface commands, queries and messages
190 * gst_rtsp_channel_new()
191 * gst_rtsp_channel_unref()
192 * gst_rtsp_channel_attach()
193 * gst_rtsp_channel_queue_message()
194 * gst_rtsp_connection_accept()
195 * GstAppSink::gst_app_sink_set_callbacks()
196 * GST_VIDEO_FORMAT_YVYU,GST_VIDEO_BUFFER_TFF,GST_VIDEO_BUFFER_RFF,GST_VIDEO_BUFFER_ONEFIELD
197 * GST_MIXER_FLAG_HAS_WHITELIST,GST_MIXER_FLAG_GROUPING,GST_MIXER_TRACK_NO_RECORD,GST_MIXER_TRACK_NO_MUTE,GST_MIXER_TRACK_WHITELIST
198 * GstAppSrc::emit-signals
199 * GstAppSrc::gst_app_src_set_emit_signals()
200 * GstAppSrc::gst_app_src_get_emit_signals()
201 * GstAppSrc::gst_app_src_set_callbacks()
202 * RTSP::gst_rtsp_connection_get_url()
203 * GstRTSPLowerTrans::GST_RTSP_LOWER_TRANS_HTTP
204 * RTSP:gst_rtsp_connection_set_tunneled()
205 * RTSP:gst_rtsp_connection_is_tunneled()
206 * RTSP::gst_rtsp_connection_set_ip()
207 * RTSP::gst_rtsp_connection_get_tunnelid()
208 * RTSP::gst_rtsp_connection_do_tunnel()
209 * RTSP::gst_rtsp_watch_reset()
213 1) Please note that decodebin2 and playbin2 API included in this release is
214 still considered unstable and WILL change in future releases. At this stage,
215 only developers or early adopters should consider using decodebin2 or playbin2
216 API embodied in their signals and properties.
218 Changes since 0.10.21:
220 * Require gettext 0.17
221 * Replace audioresample with speexresample from -bad
222 * Support new formats in RIFF: uncompressed RGB, WMA lossless, VP6
223 * Move libgstapp and elements from -bad
224 * Support color-key setting and probing for Xv properties
225 * Improve typefinding for various formats
226 * Extend audio sinks for pull-mode operation
227 * Support for more subtitle formats
228 * More development on decode2bin and playbin2
230 * Many bug fixes and improvements
232 Bugs fixed since 0.10.21:
234 * 562163 : theoraenc likely ignoring segments
235 * 562258 : rtspsrc element takes long time to error out if the addre...
236 * 561789 : [volume] deadlocks with a controller attached
237 * 554533 : [xvimagesink] allow setting colorkey if possible
238 * 567511 : colorkey in xvimagesink gets reset when element is reused
239 * 116051 : libresample doesn't handle > factor of 2 rate conversion
240 * 346218 : [audioresample] doesn't do anti aliasing
241 * 385061 : [audioresample?] investigate high CPU usage
242 * 456788 : [subparse] can't handle UTF-16 charset encoded subtitle.
243 * 525807 : [vorbisenc] vorbisenc has problems with a gnlsource that ...
244 * 546955 : gstoggmux EOS handling issue
245 * 549417 : [audioresample] unit test fails on 64bit linux
246 * 549510 : audioresample doesn't negotiate ideal caps
247 * 552237 : UTF-16 srt confuses gstreamer, misdetected as mp3
248 * 552559 : Implementation of SLAVE_SKEW in baseaudiosrc
249 * 552569 : audioresample producing strange sized buffers
250 * 552801 : audioconvert can overflow with big audio buffers
251 * 554879 : Add ability to specify format for date/time display in Gs...
252 * 555257 : Doesn't display srt subtitles saved with BOM
253 * 555319 : add FFV1 fourcc to riff-media
254 * 555607 : subrip subtitles typefind too strict
255 * 555699 : [PATCH] theoradec: prefer container's pixel aspect ratio ...
256 * 556025 : build failure in tests/icles
257 * 556066 : Last byte of FLAC image buffer chopped off
258 * 557365 : subparse check fails
259 * 558124 : [PLUGIN-MOVE] Move speexresample as audioresample2 to -base
260 * 559111 : ALSA sink hangs on USB audio device unplug while playing
261 * 559478 : does not play windows media streams correctly
262 * 559567 : `gst_base_audio_sink_sync_latency' should call `gst_base_...
263 * 561436 : videorate element add image/jpeg to caps template
264 * 561734 : playbin2 additions
265 * 561780 : Playbin2 should work without volume too
266 * 561924 : oggdemux hangs when given corrupt input via non-seekable ...
267 * 562270 : build without gdk fails
268 * 563143 : ximagesink/xvimagesink : _alloc_buffer returns non-clean ...
269 * 563174 : Implement gst_rtcp_packet_remove
270 * 563508 : [rgvolume] Unit test fails with passthrough assertions
271 * 563718 : Theora check out of date
272 * 563904 : GNOME Goal: Clean up GLib and GTK+ includes
274 API added since 0.10.21:
276 * clockoverlay::time-format
277 * GstRingBuffer:gst_ring_buffer_activate()
278 * GstRingBuffer:gst_ring_buffer_is_active()
279 * GstRingBuffer:gst_ring_buffer_convert()
280 * Add GST_TYPE_BASE_AUDIO_(SRC|SINK)_SLAVE_METHOD to the public API
281 * gst_netaddress_get_address_bytes()
282 * gst_netaddress_set_address_bytes()
284 Changes since 0.10.20:
286 * Continue playbin2 development
287 * Ogg improvements - CELT support, skeleton fixes
288 * DVD subpicture support
289 * Improved audio dithering random number generator
290 * xvimagesink/ximagesink fixes
291 * Vorbis encoding and decoding fixes
292 * Recognise Kate subtitle streams
293 * Many bug-fixes and enhancements
295 Bugs fixed since 0.10.20:
297 * 537380 : [gnomevfssrc] Doesn't handle short reads properly
298 * 538656 : xvimagesink support for autofill/colorkey property
299 * 540334 : Build fails without X in tests/examples/seek
300 * 528299 : Multiple GstMixerTracks with the same label cause problem...
301 * 530068 : Ogg Streams with Skeleton and Granulepos > 0 do not work(...
302 * 537009 : playbin2 silly typo breaks signals
303 * 537045 : decodebin2 sometimes emits 'drained' multiple times
304 * 537599 : [oggdemux] skeleton streams not skipped in ogg
305 * 537889 : [xvimagesink] colorbalance is bad
306 * 538232 : vorbisenc/vorbisdec don't work with a live source
307 * 538663 : gdppay memleak in gst_gdp_pay_reset
308 * 540215 : decodebin does not insert a queue for raw data type
309 * 540351 : [avidemux] Doesn't know about Duck DK4 ADPCM
310 * 540497 : ffmpegcolorspace is returning wrong size
311 * 541358 : cross mingw32 gcc: getaddrinfo is not in ws2_32.dll befor...
312 * 544306 : rtspsrc debug=1 segfaults with some libc
313 * 548898 : GStreamer-CRITICAL errors on seeking beyond stream borders
314 * 548913 : vorbisenc being picky about rounding errors in timestamps
315 * 549062 : Video devices aren't updated on subsequent probing.
316 * 549814 : [typefind] add application/pdf typefinder
317 * 550582 : [oggdemux] KATE streams not recognised
318 * 550638 : [typefind] Recognize some jpeg2k file types
319 * 550656 : recognize TrueSpeech in wavparse
320 * 550729 : gst-plugins-base won't compile with " -pedantic " option
321 * 552960 : tagdemux asserts and aborts on truncated files
322 * 553244 : theoraparse doesn't work at all (throws criticals and ass...
324 API added since 0.10.20:
326 * Add "index" property to GstMixerTrack to differantiate between
327 multiple mixer tracks with the same label.
329 Changes since 0.10.19:
332 * Support digest auth for RTSP
333 * Additional documentation
334 * Support DSCP QoS in multifdsink
335 * Add NV12/NV21 video buffer layouts
336 * Video scaling now bilinear by default
337 * Support more than 8 channels in audio conversions
338 * Channel mapping fixes for audioconvert
339 * Improve tmplayer and sami subtitle support
340 * Support 1x1 pixel buffers for videoscale
341 * Typefinding improvements for MPEG2, musepack
342 * Ogg/Dirac mapping updated in oggmux
343 * Fixes in ogg demuxing
344 * audiosink synchronisation and slaving fixes
345 * Support muting of the audio in playbin by selecting -1 as the audio stream
346 * Work done on playbin2 and uridecodebin
347 * Improvements in the experimental GIO plugin
349 * Handle GAP buffers in some places
350 * Various other leak and bug-fixes
352 Bugs fixed since 0.10.20:
354 * 526794 : [giosrc] totem doesn't work with some gvfs backends
355 * 510417 : [PLUGIN-MOVE] Move gio to gst-plugins-base
356 * 509125 : crash in CD Player: - playing CD - lowering/...
357 * 517813 : [audioconvert] make gap aware
358 * 302798 : [playbin] add mute property
359 * 342294 : Setting playbin property current-audio=-1 also stops the ...
360 * 398033 : [audioconvert] support more than 8 channels
361 * 419351 : [avi/a52dec] AV synchronization problems
362 * 467911 : [subparse] sami parser update
363 * 469933 : multifdsink IPv6 and diffserv TOS/TC markup
364 * 506659 : [textoverlay] rendering error when using non-standard widths
365 * 512333 : [gstvorbistag] Retrieve Ogg/Vorbis cover art as image met...
366 * 512382 : [playbin] race condition when pausing/playing multiple in...
367 * 518037 : pbutils-enumtypes.c is not included in win32/vs6/libgstpb...
368 * 521761 : gstaudioclock frozen the clock value until reaches latest...
369 * 522401 : gdpdepay doesn't validate payload CRCs
370 * 523993 : playbin2 blocks after a while when listening to a radio s...
371 * 524724 : [PATCH] [baseaudiosrc] buffer-time and latency-time do no...
372 * 525665 : Crash on Ogg/Vorbis with chain=NULL
373 * 525915 : [streamheader] Unit test fails with " gst_adapter_peek: as...
374 * 526173 : [typefinding] fails to detect mpeg video stream whereas m...
375 * 529018 : gst_ogm_parse_stream_header creates fraction value with w...
376 * 529500 : [videotestsrc] support for NV12 and NV21
377 * 529546 : [Playbin] Memory leak in streaminfo handling
378 * 530068 : Ogg Streams with Skeleton and Granulepos > 0 do not work(...
379 * 530531 : [typefinding] bad read in mpeg_video_stream_type_find
380 * 530719 : gst_video_calculate_display_ratio fails when playing Ogg ...
381 * 530962 : [subparse] parses only every second line of TMPlayer subt...
382 * 532454 : [NV12/NV21] videotestsrc and ffmpegcolorspace don't play ...
383 * 533087 : GstRTSPTransport kept opaque in docs
384 * 533817 : [audioconvert] Can't use default 7 channel layout / only ...
385 * 534071 : Gdppay memleak
386 * 534331 : race in decodebin when changing states while the internal...
387 * 535356 : vorbisdec doesn't support 8 channels
388 * 536475 : gdppay memleak and possible crash
389 * 536521 : Refcounting errors in playbin
390 * 536874 : Build failure on windows
391 * 532166 : [ffmpegcolorspace] support NV12 format
392 * 533617 : [audioconvert] Produces silence when converting 1/2 chann...
393 * 536848 : [giosrc] Doesn't handle short reads properly
394 * 536849 : [giosrc] Very slow doing any playback
395 * 518082 : [alsamixer] playback volumes overwritten by capture volum...
396 * 435633 : [PATCH] videorate not (fully) segment aware; causes frame...
397 * 532364 : tcpclientsrc broken in 0.10.19
398 * 533075 : gst_rtp_buffer_compare_seqnum doesn't do what it says
399 * 533265 : [cddabasesrc] Sound Juicer cut a sector when ripping a track
401 API additions since 0.10.20:
403 * decodebin2::sink-caps property
404 * giosrc::file property
405 * giosink::file property
406 * gst_base_audio_src_set_slave_method()
407 * gst_base_audio_src_get_slave_method()
408 * GstAudioClock::gst_audio_clock_reset()
409 * GstBaseAudioSrc:actual-buffer-time property
410 * GstBaseAudioSrc:actual-latency-time property
411 * gst_audio_check_channel_positions()
412 * add gst_tag_image_data_to_image_buffer()
413 * add gst_tag_list_add_id3_image()
414 * add GST_TAG_IMAGE_TYPE_NONE enum value
416 Changes since 0.10.18:
418 * Handle EAGAIN when polling sockets in rtspconnection
420 Changes since 0.10.17:
422 * Experimental GIO plugin
423 * Continued playbin2 development
425 * Better network element support on Windows
426 * Various other bug-fixes and improvements
428 Bugs fixed since 0.10.17:
430 * 509637 : [API] [basertpaudiopayload] add _set_samplebits_options()
431 * 510229 : [gnomevfssrc] HTTPS support
432 * 511478 : [rtpbuffer] add gst_rtp_buffer_set_extension_data function
433 * 511810 : [RTSP] Uses MT-unsafe gmtime() function
434 * 512899 : [alsa] gstalsasink.c:527: warning: 'snd_pcm_sw_params_set...
435 * 513167 : Fix compiler warning due to disabled signals in mixertrac...
436 * 514307 : [playbin] warning in nautilus, volume element can't be cr...
437 * 514623 : Ogg Theora video slow
438 * 514937 : Correct initialization of hints in is_multicast_address()
439 * 515654 : xvimagesink doesn't build with --disable-xshm
440 * 516246 : [alsasink] handle negative delay from snd_pcm_delay
441 * 517420 : typefind: add h264 elementary stream discovery
442 * 517991 : problems with configure file depending on GCC compiler
443 * 518039 : libgstrtsp MSVC 6.0 compile error
444 * 518162 : [subparse] handle italic text starting with " / " with Micr...
445 * 518940 : [playbin2] make _get_*_tags() match vfuncs prototype in c...
446 * 519906 : [API] add GstMixerOptions::get_values vfunc
447 * 519916 : [API] add mixer-changed and options-list-changed messages
448 * 520523 : [API] Unreviewed changes to ringbuffer API
449 * 521743 : libgstnetbuffer.def exports not up to date
450 * 522625 : [video] gst_video_format_parse_caps() broken for RGBA for...
451 * 523054 : gstbasesrc crashes when called from typefind helpers
452 * 511825 : [RTSP] compiler warning on FreeBSD
453 * 520300 : [alsasrc] provide-clock=false messes up buffer durations
455 API added since 0.10.17:
457 * GstRTPBuffer:gst_rtp_buffer_set_extension_data()
458 * add GST_VIDEO_FORMAT_Y41B and GST_VIDEO_FORMAT_Y42B.
459 * add GstMixerOptions::get_values vfunc (#519906)
460 * add gst_mixer_options_list_changed(), gst_mixer_mixer_changed() and
461 gst_mixer_message_parse_options_list_changed(). Fixes #519916.
462 * gst_base_rtp_audio_payload_set_samplebits_options()
463 * GstNetBuffer::gst_netaddress_equal
465 Changes since 0.10.16:
467 * Work-around ABI breakage due to unfortunate use of the
468 GST_DISABLE_DEPRECATED macro
469 * Export 2 missing functions needed for bindings in the win32 build
470 * Initialise the GstRingBuffer GType from a thread-safe context
472 Bugs fixed since 0.10.16:
474 * 511825 : [RTSP] compiler warning on FreeBSD
475 * 513018 : crash in Volume Control: I typed my password at t...
476 * 512334 : g_critical() when using GstAudioFilter & GST_DEBUG
478 Changes since 0.10.15:
480 * Handle newer Theora granule-pos semantics
481 * Introducing first alpha version playbin2 - the upcoming successor to
483 * Fixes in playbin handling of stream-switching
484 * New API for uniform handling of raw-video format buffers.
485 * Improvements for RTSP/RTP handling
486 * RIFF lib additions for VC-1 and AVC1 fourccs
487 * Many other bug-fixes and improvements
489 Bugs fixed since 0.10.15:
491 * 506132 : Review of changes in video/video.h
492 * 320984 : [oggdemux] cannot handle multiple chains
493 * 373011 : [playbin] throws error when switching off subtitles
494 * 436756 : Intermittent crashes in Pidgin in audioclock g_type_class...
495 * 462740 : [streamselector] patch to improve default stream selection
496 * 486840 : [alsamixer] use _all variants when setting the mixer
497 * 497964 : theoraenc test fails
498 * 498228 : gst-plugins-base-0.10.15 does not compile on FreeBSD (Gen...
499 * 499697 : Provide better pkg-config files
500 * 502497 : [subparse] SubRip subtitles starting from 0 not recognised
501 * 503440 : The control sockets used by gstrtspconnection.c are never...
502 * 503930 : [cdda] warning: 'eos' may be used uninitialized in this f...
503 * 506928 : [alsamixer] add " PCM " as master fall back for cards that ...
504 * 508138 : [decodebin] does not error out if pad activation fails
505 * 509762 : missing file in win32/MANIFEST
506 * 511274 : gst_rtp_buffer_get_extension_data is returning FALSE when...
507 * 496731 : [PATCH] xvimagesink leaks memory if initialization fails
508 * 496761 : [PATCH] RTSP message leaks memory when uninitialized
509 * 500763 : SIGSEGV while playing ogg audio file
511 API additions since 0.10.15:
513 * New GstVideoFormat API and helper functions in libgstvideo
514 * gst_base_audio_sink_set_provide_clock()
515 * gst_base_audio_sink_get_provide_clock()
516 * gst_base_audio_sink_set_slave_method()
517 * gst_base_audio_sink_get_slave_method()
518 * gst_base_audio_src_set_provide_clock()
519 * gst_base_audio_src_get_provide_clock()
521 Changes since 0.10.14:
523 * RTP/RTSP/RTCP/SDP support improved
524 * New FFT support library libgstfft, based on Kiss FFT
525 * New formats supported in volume and audiotestsrc
526 * Fixes in audiorate and videorate
527 * Audio capture fixes
528 * Playbin and decodebin fixes
529 * New tagdemux base class for ID3/APE style tag readers
530 * Fix a nasty crash in the X sinks on shutdown
532 * Add support for multichannel WAV files.
533 * Preserve channel layout information when up/down-mixing.
534 * Many bug-fixes and improvements
536 Bugs fixed since 0.10.14:
538 * 475395 : decodebin2 leaks request-pads
539 * 475451 : [decodebin2] leaks ghostpad
540 * 378770 : [xvimagesink] race condition in event thread?
541 * 407282 : [decodebin2] autoplug-sort signal has GList ** parameter
542 * 430677 : [audioconvert] does not preserve channel positions when f...
543 * 442654 : [volume] controller bypassed by default
544 * 445529 : [volume] support for 24/32-bit audio/x-raw-int
545 * 446766 : return code for gst_base_rtp_payload_audio_handle_event()
546 * 451970 : Subparse requires HTML parser
547 * 453650 : [audiobasesrc] two alsasrcs do not work in one pipeline
548 * 459334 : [textoverlay] expose pango line alignment property
549 * 459585 : [basertpdepayload] api without namespace
550 * 460422 : [audiotestsrc] Add support for float and double output
551 * 462805 : [alsa] compilation fails with gcc 4.2
552 * 462979 : Add 'silent' property to GstTimeOverlay
553 * 463215 : [audioconvert] compile errors
554 * 464320 : [PATCH] gst-plugins-base-0.14 does not build for win32
555 * 464666 : [playbin] QT trailer hangs in preroll with decodebin2
556 * 464690 : Add connection-speed property to uridecodebin element
557 * 465015 : [playbin] Not removed probes causes deadlocks in streamin...
558 * 465028 : some warnings with mingw
559 * 467667 : GST_FRAMES_TO_CLOCK_TIME() and GST_CLOCK_TIME_TO_FRAMES()...
560 * 468129 : [basertpaudiopayload] event handler returns the wrong value
561 * 468619 : New library gstfft: FFT library for integer and float typ...
562 * 470456 : [API] add gst_missing_*_installer_detail_new()
563 * 470766 : [ssaparse] line breaks in SSA subtitle parser
564 * 471067 : Make the SDP code useable for generating SDP descriptions
565 * 471194 : [rtpbuffer] RTP headers are wrong for win32
566 * 473097 : [baseaudiosink] gstreamer-properties hangs when testing s...
567 * 474384 : gstrtsp-enumtypes.c and .h needed for win32
568 * 474880 : [xvimagesink] [ximagesink] leaking buffer caps reference
569 * 475731 : rtspconnection is able to read incomplete messages
570 * 483620 : All Rtp buffers are discarded -- gst_rtp_buffer_get_payl...
571 * 484989 : memleak, not unrefed caps for gstbasertppayload.c
572 * 489010 : Please change default channel order for WAVE_EXT-less .wa...
573 * 491722 : [playbin] regression: crash with external subtitles
574 * 492098 : [GstFFT] Broken scaling
575 * 492114 : Build issues on Windows/MSVC
576 * 492306 : compilation errors with MinGW
577 * 492813 : Missing symbols in libgstrtp.def
578 * 493986 : Build issues on Windows (missing symbols)
579 * 494346 : pre-release vs6 patch
580 * 496548 : Including malloc.h breaks macos build
581 * 496724 : DSW file references non-existent DSP files
582 * 464079 : audiotestsrc doesn't respond to conversion queries properly
583 * 442065 : floatcast.h includes config.h and might break other apps
584 * 466717 : gst_event_new_new_segment_full:assertion `start < = stop' ...
585 * 485753 : Decodebin2 deadlocks when nulling pipeline during typefind
586 * 464028 : Move connection-speed from playbin to playbasebin
588 API added since 0.10.14:
590 * GstTagDemux base class for simple tag demuxers
591 * GstBaseAudioSrc::provide-clock property
592 * gst_rtcp_ntp_to_unix()
593 * gst_rtcp_unix_to_ntp()
594 * gst_rtp_buffer_get_header_len()
595 * gst_rtp_buffer_get_extension_data()
596 * gst_rtp_buffer_compare_seqnum()
597 * gst_rtp_buffer_ext_timestamp()
598 * gst_rtcp_packet_sdes_copy_entry()
599 * gst_install_plugins_supported()
600 * gst_missing_*_installer_detail_new() convenience API
601 * gst_rtsp_connection_poll()
602 * GstTextOverlay::line-alignment property
604 Changes since 0.10.13:
606 * Audio dither and noise-shaping when reducing bit-depth
607 * RTSP and SDP helper libraries added
608 * Experimental buffering element "queue2" now supports pull-mode
609 and file-based buffering.
610 * Support for more 32-bit video pixel layouts
611 * Various fixes and improvements
613 Bugs fixed since 0.10.13:
615 * 380625 : [x*imagesink] add 'handle-expose' property
616 * 385527 : oggmux sometimes gets DELTA flag on output wrong near start
617 * 402076 : videoscale 4-tap method broken for downscaling
618 * 437169 : [xvimagesink] add property to disable Xv double-buffering
619 * 441264 : queue2 support to do buffering on a file
620 * 442553 : [v4lsrc] doesn't output segments in GST_FORMAT_TIME
621 * 442557 : [videorate] doesn't handle latency queries
622 * 442944 : Audiotestsrc can overflow on seeks
623 * 444523 : [queue2] Pull mode support
624 * 444630 : Compilation error with fsseko (from gstqueue2.c) -- unabl...
625 * 445505 : [queue2] It does not work in pull mode with oggdemux
626 * 446551 : [queue2] Buffering is not working properly if it is set t...
627 * 446572 : [queue2] Division by zero
628 * 446972 : warning when compiling gstoggdemux.c
629 * 449156 : Regression in CVS for decodebin2
630 * 450875 : Missing files in po/POTFILES.in
631 * 451707 : [tag] UTF-8 in ID3v1 tag not correctly decoded
632 * 451908 : [ffmpegcolorspace] regression: doesn't accept GST_VIDEO_C...
633 * 454264 : Playbin fails to " play " image url after a movie url
634 * 456656 : [API] Addition of audio buffer clipping function to gstaudio
635 * 460978 : gst_audio_buffer_clip outputs warnings
636 * 152864 : [PATCH] GstAlsaMixer doesn't support signals
637 * 360246 : [audioconvert] Optionally apply dithering
638 * 394061 : Add support for Subviewer subtitles
639 * 420326 : Base payloader class has wrong property types and ranges
640 * 451145 : [vorbisdec] errors out on 0-sized packets
641 * 459204 : [PATCH] [playbin] gst_play_base_bin_get_streaminfo_value_...
643 API added since 0.10.13:
645 * RTSP and SDP libraries added
646 * gst_rtsp_base64_decode_ip
647 * Add buffer clipping function gst_audio_buffer_clip for raw audio
648 buffers. Fixes #456656.
649 * gst_mixer_get_mixer_flags
650 * gst_mixer_message_parse_mute_toggled
651 * gst_mixer_message_parse_record_toggled
652 * gst_mixer_message_parse_volume_changed
653 * gst_mixer_message_parse_option_changed
654 * GstMixerMessageType
657 Changes since 0.10.12:
658 * Many fixes and improvements
659 * RTP and RTCP support improved
661 Bugs fixed since 0.10.12:
663 * 339838 : [audioconvert] support floats with non-native endianness
664 * 393975 : closing x/xvimagesink window crashes gst-launch
665 * 405072 : [API] add gst_tag_freeform_string_to_utf8()
666 * 413799 : [subparse] add support for MPL2 format
667 * 414645 : GstMixerTrack should make untranslated label available
668 * 420079 : [audioconvert] Uses biased rounding which results in dist...
669 * 420578 : [subparse] add more colour map in sami parser
670 * 421834 : videorate breaks on dimension changes
671 * 423051 : Vorbis tags of type double use locale-dependent formatting
672 * 423055 : Verify ReplayGain vorbistag processing in libs/tag testsuite
673 * 425455 : Decodebin2 leaks pads
674 * 426250 : GstPlayBaseBin leaks streaminfo objects
675 * 428187 : Rtp base depayloader class doesn't send new_segment after...
676 * 431672 : gst_base_rtp_audio_payload_push() should take object of i...
677 * 432362 : [ximagesink] doesn't build if XShm is not available
678 * 432755 : [videorate] leaks buffer if flow != OK
679 * 432984 : [baseaudiosrc] misleading warning message when dropping s...
680 * 433888 : [theoradec] does not generate a perfect stream
681 * 436562 : Theoradec doesn't work well with gnonlin
682 * 438840 : [theoradec] does not compile with old version of libtheora
683 * 440997 : [gstriff] Doesn't handle width!=depth files with audio/x-...
684 * 441295 : audioconvert doesn't build on VS6
685 * 442024 : regression in playbin buffering
686 * 350299 : [playbin] " Internal data flow error " opening movie with s...
687 * 410039 : totem crashed with SIGSEGV in new_decoded_pad_full()
688 * 340842 : do latency calculation for live sources
689 * 341078 : RB does not play beyond initially downloaded podcast file
690 * 414496 : [id3demux, id3v2mux] Add support for GST_TAG_MUSICBRAINZ_...
692 API additions since 0.10.12:
694 * add gst_tag_freeform_string_to_utf8()
695 * GstRTPBuffer::gst_rtp_buffer_default_clock_rate()
696 * GstBaseAudioSink::slave-method property
697 * add "min-ptime" property to RTP base audio payloader
698 * gst_base_rtp_audio_payload_push()
699 * gst_base_rtp_audio_payload_get_adapter()
700 * GstMixerTrack::untranslated-label property
702 Changes since 0.10.11:
704 * New API for on-demand plugin installation
705 * Xv thread-safety and configuration enhancements
706 * decodebin2 improvements
707 * Support more raw audio format conversions
708 * Improvements in Ogg support
709 * AudioFilter base class ported to 0.10
710 * Fixes for subtitles
711 * Latency/live-playback support for Alsa
712 * Lots of bug fixes and improvements
714 Bugs fixed since 0.10.11:
716 * 398721 : No video in .ogm files with decodebin2
717 * 339837 : [audioconvert] support for 64-bit float audio
718 * 341524 : [decodebin] can't handle decoders with always src pads wi...
719 * 352069 : Add de.po German translation
720 * 363379 : [oggmux] doesn't detect EOS on all sinkpads
721 * 378436 : [oggdemux] rhythmbox crash on fast clicking on rating in ...
722 * 380342 : Totem does not play mp3 files when lyrics are present
723 * 383195 : [cddabasesrc,basertpaudiopayload] compile errors with gcc...
724 * 383198 : totem crashed to gst_xvimagesink_update_colorbalance
725 * 384008 : [xvimagesink] accesses - > xwindow outside locks
726 * 384060 : gst_xoverlay_set_xwindow_id() causing lockups with x(v)im...
727 * 387138 : x input events processing in sinks with xoverlay interfac...
728 * 390063 : Documentation typo
729 * 390076 : add xv adaptor and port properties in xvimagesink element.
730 * 391365 : [oggdemux] internal stream error on OggFlac
731 * 392070 : [vorbis] GST_TAG_LOCATION not mapped
732 * 392393 : [API] add libgstbaseutils library for missing plugins mes...
733 * 396042 : mpeg4 video typefinder loops endlessly on quicktime redirect
734 * 396835 : audioconvert/audioresample combination causing buffer of ...
735 * 397673 : [patch] XIOError caught in x[v]imagesink.c
736 * 397810 : [typefinding] .vob file: could not determine type of stream
737 * 398110 : [theoraenc] GLib failed to allocate 3080991032 bytes on g...
738 * 399340 : Crash in the oggdemux plugin when trying to play a specia...
739 * 401029 : [playbin] rapidly changing visualisation freezes
740 * 401072 : Move libgimme-codec helper functions to GStreamer
741 * 402505 : visualisations don't work for some samplerates
742 * 407811 : decodebin2 hang on HD clip
743 * 409683 : Crash with Decodebin2
744 * 410396 : not reading " DATE " tags from Flac files
745 * 410963 : Fails to build with -z defs
746 * 357503 : [suparse] wrong timing with microdvd subtitles
747 * 393310 : [pango] localtime_r does not exist in MinGW
748 * 397207 : Test failure w/ HP-UX 11.11 & native compiler
749 * 399948 : [textoverlay] leaks upstream events if textpad unlinked
750 * 403963 : GstAudioFilter base class broken
751 * 404512 : [videoscale] floating point exception on 1x1 video
752 * 405020 : [alsa] probing the device-name doesn't seem to work corre...
753 * 408278 : [videorate] memory leak
754 * 410772 : Crash copying a GstNetBuffer
755 * 401118 : [visual] error if width not a multiple of 4
756 * 405451 : [alsasink] deadlocks when disconnecting USB Sounddevice
758 API additions since 0.10.11:
761 * GST_VIDEO_SINK_CAST()
762 * gst_pb_utils_add_codec_description_to_tag_list()
763 * gst_pb_utils_get_codec_description()
764 * gst_pb_utils_get_source_description()
765 * gst_pb_utils_get_sink_description()
766 * gst_pb_utils_get_decoder_description()
767 * gst_pb_utils_get_encoder_description()
768 * gst_pb_utils_get_element_description()
769 * gst_pb_utils_init()
770 * gst_install_plugins_context_new()
771 * gst_install_plugins_context_set_xid()
772 * gst_install_plugins_context_free()
773 * gst_install_plugins_async()
774 * gst_install_plugins_sync()
775 * gst_install_plugins_return_get_name()
776 * gst_install_plugins_installation_in_progress()
777 * gst_missing_uri_source_message_new()
778 * gst_missing_uri_sink_message_new
779 * gst_missing_element_message_new
780 * gst_missing_decoder_message_new
781 * gst_missing_encoder_message_new
782 * gst_missing_plugin_message_get_installer_detail
783 * gst_missing_plugin_message_get_description
784 * gst_is_missing_plugin_message
786 Bugs fixed since 0.10.10:
788 * 360552 : [riff] [avi] extracts non-UTF8 metadata
789 * 365501 : [x/xvimagesink] race condition when creating first image ...
790 * 339366 : [playbin] hangs if suburi file type cannot be determined
791 * 355914 : libvisual causes xvimagesink: assertion `GST_CAPS_REFCOU...
792 * 363118 : gst_riff_create_video_caps() should also store variant in...
793 * 363607 : xvimagesink xwindow_draw_border() slowness
794 * 336301 : [playbin] can't handle RTSP source
795 * 337026 : oggmux doesn't set EOS properly
796 * 337031 : vorbisdec outputs too much data
797 * 340049 : New BaseRTPAudioPayloader class to -base
798 * 348264 : Theora encoding, Ogg muxing don't handle discontinuities
799 * 354773 : xvimage assumes that XV_COLORKEY can be set in RGB888 format
800 * 355917 : libvisual plugin is broken
801 * 355935 : multifdsink doesn't allow setting maximums (soft, hard) i...
802 * 357038 : [ffmpegcolorspace] RGBA handling broken
803 * 357215 : [playbin] buffering notification not quite right yet
804 * 357289 : [riff] riff parser can't detect aac audio stream
805 * 357404 : [playbin] Linking can fail silently
806 * 357531 : [subparse] problem if markup is not closed
807 * 357577 : [playbin] regression: buffering still images broken
808 * 357591 : Avoid compiler warning with uclibc and -Werror
809 * 357613 : XvStopVideo in xvimagesink
810 * 357800 : [libvisual] doesn't pass audio data to libvisual 0.4.0 co...
811 * 359580 : tcpserversink and dataprotocol assert for multipart streams
812 * 361095 : Fixes compiling with forte: warning clean up (part 3)
813 * 361456 : [basertppayload] Memory leak
814 * 361634 : sink- > ringbuffer NULL in BaseAudioSink's setcaps()
815 * 361984 : [subparse] doesn't accept .srt file that doesn't start wi...
816 * 366334 : [PATCH] Windows vs8 fixes
817 * 368273 : Using the remove signal on multifdsink is not threadsafe
818 * 368310 : include file gstbasertpaudiopayload.h not included for r...
819 * 369482 : [typefind] MPEG system streams get recognized as mp3 files
820 * 370092 : [PATCH] Decodebin v2 : Implementation
821 * 377183 : regression: no eos when playing ogg vorbis files
822 * 381219 : bad debugging code left in audiorate
823 * 382223 : [decodebin] more delayed linking
824 * 382269 : Typefind detects mpeg video clip as audio/mpeg
825 * 335635 : Add an Ogg/Vorbis retagging element
826 * 341681 : [textoverlay] flickering with continuously timestamped text
827 * 342228 : [alsa] Recognize " Front " as a Master channel
828 * 357330 : [subparse] some sami parser minor but enhanced patch
829 * 357532 : [gsttag] vorbistag doesn't handle dates that include time...
830 * 359237 : [typefinding] doesn't recognize XML files shorter than 25...
831 * 362845 : [subparse] add support for tmplayer format
832 * 357977 : [videorate] new segment start is not respected
833 * 364812 : [PATCH] oggmux release pad does not remove pad
834 * 364856 : pngenc stride problems
835 * 372507 : Mac build fixes
837 API added since 0.10.10:
839 * playbin::queue-min-threshold property.
840 * GstVideoOrientation interface
841 * gst_base_rtp_depayload_push_ts
842 * gst_base_rtp_depayload_push
843 * Add dropped_buffers to multifdsink's get-stats GValueArray
844 * gst_ring_buffer_commit_full
846 Changes since 0.10.9:
848 * New elements: gdppay, gdpdepay
850 Bugs fixed since 0.10.9:
852 * 343787 : The adder cannot handle when multiple elements tries to l...
853 * 336075 : ALSA emu10k1 mixer tracks are wrongly classified as playb...
854 * 349105 : crash with playbin and resizing screen
855 * 342494 : [v4l] Query " device-name " even if device is not open
856 * 342680 : [adder] seeking with multiple ogg files fails to work
857 * 345188 : [alsa] can't handle more than 8 channels
858 * 347091 : converting vorbis comments to GstTagLists is lossy
859 * 348157 : Changed " Change Device " menu behaviour in gnome-volume-co...
860 * 348916 : [typefind] add multipart/x-mixed-replace typefinder
861 * 350157 : [riff] riff parser can't detect dts audio stream
862 * 350655 : [oggdemux] should process seeking queries
863 * 350900 : [adder] should not clamp floating point values
864 * 351426 : API: add gst_tag_parse_extended_comment
865 * 351502 : g_value_set_string leaks
866 * 351742 : [vorbisenc] discontinuity detection too sensitive, might ...
867 * 353658 : [videotestsrc] doesn't round strides correctly for YVYU
868 * 354594 : multifdsink doesn't work reliably with sync-method = 'nex...
869 * 351790 : [ogmparse] crash parsing video stream on x86-64
870 * 140139 : [avidemux] can't play broken avi with ogg (not vorbis) au...
871 * 347783 : [PLUGIN-MOVE] GDP elements should be moved
872 * 347918 : Internal data flow error in udpsrc
873 * 349656 : jitterbuffer in GstBaseRtp fails to handle rtp seqnum rol...
874 * 350784 : element alsamixer doesn't respect asoundrc
875 * 351308 : [netbuffer] build fails with gkt-doc critical warnings
876 * 353234 : audiorate preserves DISCONT on buffers
877 * 353912 : Add cmml caps to oggmux
879 API added since 0.10.9:
881 * gst_rtp_buffer_get_payload_subbuffer()
882 * gst_tag_parse_extended_comment()
883 * GstPlayBin::connection-speed
884 * GstTheoraParse::synchronization-points
885 * GST_AUDIO_CHANNEL_POSITION_NONE
887 Changes since 0.10.8:
889 * Parallel installability with 0.8.x series
890 * Threadsafe design and API
892 * Support for images in tags
893 * Playback improvements
894 * Gnomevfssrc now supports burn:// uris
895 * Videoscale now supports more RGBA formats
896 * Multifdsink improvements
897 * Testsuite can now generate coverage information
899 Bugs fixed since 0.10.8:
901 * 347296 : Problems with clocks on alsasrc hangs the application
902 * 347295 : [vorbisdec] Pushes before being initialized
903 * 329798 : [playbin] doesn't always give correct error message for m...
904 * 342085 : [alsasink] doesn't set buffer-time correctly
905 * 342789 : [audioresample] doesn't clear state when stopped, causing...
906 * 343303 : [subparse] workaround for bad entities in sami parser
907 * 343385 : [gnomevfs] add support for burn:// URIs
908 * 343500 : [riff] gst_riff_parse_strf_vids() can't parse extra data.
909 * 343699 : oggmux leaks
910 * 344503 : [subparse] parse font face property in sami parser.
911 * 345131 : [PATCH] videoscale support for 32-bit RGB-formats
912 * 345206 : [textoverlay] crash with non-UTF8 input
913 * 345225 : [theoradec] Clipping for exact seeking
914 * 345641 : [API] [libgsttag] add enums for image tag type
915 * 345879 : [riff] won't play a .wmv file with WMVA video stream
916 * 346581 : [typefinding] recognise text/html
917 * 347221 : [audioconvert] channel remapping does not work right
918 * 347304 : Massive leaks with xvimagesink
919 * 346527 : alsasrc get_range does not respect requested size
921 Changes since 0.10.7:
923 * alsasink probing fixes
924 * xvimagesink error reporting fixes
927 * vorbis multichannel fixes
928 * multifdsink streamheader fixes
930 Bugs fixed since 0.10.7:
932 * 169936 : [subparse] support for SAMI subtitles
933 * 315312 : Gstreamer Xv uses RGB instead of YUV.
934 * 334002 : video4linux shouldn't depend on X in configure script
935 * 336881 : [libvisual] additional support for libvisual-0.4
936 * 337544 : [xvimagesink] Internal Error when image is too large
937 * 339520 : [subparse] add " encoding " property
938 * 340909 : [alsasink] can't enable spdif output
939 * 341542 : some users have an assertion failed: (GST_VIDEO_SINK_WIDT...
940 * 341562 : audioconvert doesn't list formats in order of preference
941 * 341696 : audioconvert crashes if converting from a format with no ...
942 * 341719 : bisection algorithm in ogg doesn't bisect in some cases
943 * 341732 : [alsasink] doesn't query supported sample rates
944 * 341873 : [alsasink] minor memory leak, uses unprotected static var...
945 * 342143 : [subparse] sami parser needs to escape characters
946 * 342181 : [alsa] add property probe interface to alsasink and alsasrc
947 * 342268 : [playbin] add 'subtitle-encoding' property
948 * 342345 : [riff] Elephant's Dream AVI does not play, JUNK chunk bef...
949 * 342566 : Building without GTK+ fails
950 * 343397 : H.264/AAC movie deadlocks with totem in gstreamer code, p...
951 * 339935 : [adder] dead-locks when adding sink pads in PAUSED state
953 Changes since 0.10.6:
955 * typefind improvements
956 * bug-fixes in textoverlay, audioconvert, videotestsrc,
957 multifdsink and audio source/sink base classes
958 * Ice-cast metadata support has moved from gnomevfssrc to the
959 icydemux element in gst-plugins-good
960 * audioresample now supports floating point samples
961 * Adder element fixes.
962 * Fixes for network playback and audio resampling in playbin
964 Bugs fixed since 0.10.6:
966 * 340060 : [adder] handle newsegment events properly
967 * 340375 : [API 0.11] [patch] typefind to differentiate between mp4 ...
968 * 339405 : [textoverlay] can't display '\n' character
969 * 338657 : [patch] adder should send events from src-pad to all sink...
970 * 338919 : [patch] alsasink should also query witdh capabilities fro...
971 * 301759 : [audioresample] float audio support (for OSX audio sinks)
972 * 331901 : [videotestsrc] framerate=0/1 gives assertion error
973 * 333657 : Replacing icy demuxing in gnomevfssrc
974 * 336339 : [audioresample] should support width != 16
975 * 338718 : [patch] [audioconvert] correctly clip float samples > 1.0
976 * 338778 : [patch] Bad audio with ASX files
977 * 338991 : [patch] Videoscale doesn't pass on pixel-aspect ratio
978 * 339574 : [patch] Race condition in multifdsink can lead to spuriou...
979 * 339786 : [typefinding] wavpack typefinding doesn't always work
980 * 340369 : [volume element] " volume " property range insufficient
981 * 340379 : [playbin] doesn't insert audioresample, causes problems w...
982 * 340392 : Problem with internal-decodebin
983 * 341160 : [multifdsink] client_status enum has an uninitialized nick
984 * 341182 : Accessing playbin's streaminfo property from high languag...
985 * 341432 : [playbin] automatically get icecast metadata requiring ic...
986 * 341542 : some users have an assertion failed: (GST_VIDEO_SINK_WIDT...
987 * 341557 : Map GST_TAG_IMAGE < = > ID3v2 APIC tag
989 API added since 0.10.6:
991 * client-fd-removed signal added to multifdsink
992 * stream-info-value-array property added to playbin
993 * gst_video_calculate_display_ratio() in libgstvideo
995 Changes since 0.10.5:
997 * QoS in sinks and transform elements
998 * Needs GStreamer 0.10.5 for new GstBaseSink::async_playback() vmethod
999 * added theoraparse element
1001 Bugs fixed since 0.10.5:
1003 * 313136 : [playbin] hang while playing truncated ogg file
1004 * 172848 : [subparse] subtitles with special chars are displayed as ...
1005 * 305279 : [riff] uncompressed AVIs with 24bpp don't work
1006 * 320765 : [ffmpegcolorspace] make win32+msvc compliant, don't use _...
1007 * 323852 : Disable tests/icles on platforms that do not have X
1008 * 325653 : build errors compiling audioresample on win32(vs7)
1009 * 327357 : gst-plugins-base fails to compile with GCC 4.1
1010 * 334620 : [gnomevfssrc] fails to connect to icecast streaming servers
1011 * 334822 : [ffmpegcolorspace] YVU9 support
1012 * 335028 : [typefinding] ID3 v1 tag is not recognized with mp3-in-wa...
1013 * 335365 : inefficient use of GList in gst-plugins-base
1014 * 336190 : [gnomevfssink] should accept non-URI filenames as " location "
1015 * 336194 : [gnomevfssrc] some minor memory leaks
1016 * 336477 : plugins need better/univied descriptions
1017 * 336617 : Unable to recognise MPEG TS stream
1018 * 337548 : Memory leaks in basertpdepayload
1019 * 337945 : [oggdemux] segment stop position ignored
1020 * 338419 : Regression in the handling of files with multiple audio/s...
1021 * 338897 : Videoscale crashes as part of DVD to Ogg transcoding
1022 * 339013 : [videorate] Goes into an infinite loop
1023 * 339047 : [riff] handle H264 fourcc in addition to h264
1024 * 339212 : ISO file typefinding regression
1025 * 330748 : deadlock in base audio sink on playing- > paused state change
1027 Bugs fixed since 0.10.4:
1029 * 334216 : [gnomevfssrc] won't open some media on NFS mounts any longer
1030 * 334226 : typefindfunctions plugin crashes on PPC on registration
1032 Changes since 0.10.3:
1034 * (Experimental) QoS support
1035 * oggmuxer now creates 100% valid streams for Theora, Vorbis and Speex
1036 * documentation updates
1037 * better support for subtitles (seeking)
1039 Bugs fixed since 0.10.3:
1041 * 310202 : [subtitles] < i > < /i > tags and others should be supported i...
1042 * 312439 : XVideo output doesn't work on remote displays (probably r...
1043 * 321271 : audio output is truncated at EOS
1044 * 321650 : Can't decode this ogm file
1045 * 325732 : [oggdemux] problem when seeking to time less than 4s with...
1046 * 325972 : [typefinding] doesn't recognise this mp3
1047 * 326720 : [alsasink] doesn't support more than 2 channels anymore
1048 * 330711 : [ffmpegcolorspace] problems with palettized RGB (fencount...
1049 * 330789 : gstbaseaudiosink causes noise on seeking
1050 * 330888 : Fix build with gcc 2.95 (again)
1051 * 331295 : gnomevfssink doesn't respect umask when creating files
1052 * 331526 : 3GP type detection is too simple
1053 * 331678 : Decodebin is not reusable within a single pipeline (as in...
1054 * 331690 : playbin won't play my last.fm stream
1055 * 331763 : [alsamixer] unmute sets the volume to 100%
1056 * 331765 : [alsamixer] mixer applet slider doesn't want to move from...
1057 * 331903 : [videorate] doesnt handle input caps of framerate=0/1 sanely
1058 * 332778 : [ogmparse] " Already an existing pad " WARNING
1059 * 332964 : random crashes in mp3_type_find
1060 * 333254 : theora encoder does not set IN_CAPS flag properly
1061 * 333352 : [gnomevfssink] reports disk full as generic error
1062 * 333488 : Allow for palette < 256 colours in AVI files
1063 * 333510 : [PATCH] Fix gst_pad_new_from_template (gst_static_pad_tem...
1064 * 333545 : [riff] set depth on wma caps to make asfdemux and pitfdll...
1065 * 333663 : [patch] unref the result of gst_pad_get_parent
1066 * 333900 : [typefind] cannot play a particular mp3 file
1067 * 334112 : variable not initialized
1068 * 334129 : Disable frame dropping for now
1069 * 317038 : use default channel layout if none is specified in multic...
1070 * 319340 : [cdparanoia] uncorrected-error signal never fired
1072 API added since 0.10.3:
1074 * GstTextOverlay::halignment
1075 * GstTextOverlay::valignment
1077 Changes since 0.10.2:
1079 * typefind improvements
1080 * Ogg decoding and encoding fixes
1081 * Improved audio and video sink classes
1082 * Bug and leak fixes
1083 * Improved video scaling
1084 * On-the-fly visualisation switching
1087 Bugs fixed since 0.10.2:
1089 * 330244 : gsttextoverlay.c:895: 'struct _GstCollectData' has no mem...
1090 * 324000 : [playbin] post error or message on unknown input
1091 * 153004 : [typefind] can't identify mp3 file with one single mpeg f...
1092 * 323874 : [playbin] leaks sinks and threads when using gconfaudiosink
1093 * 324626 : ffmpegcolorspace support for fourcc " UYVY "
1094 * 326447 : check that all elements in -base pass queries they can't ...
1095 * 328263 : Fix build with gcc 2.95
1096 * 328279 : [decodebin] timeout issue when pre-rolling
1097 * 329326 : Fix oggmux removing pads from collect pads
1099 Changes since 0.10.1:
1101 * ported gnomevfssink, cdparanoia
1102 * New library and base class: GstCddaBaseSrc
1103 * ported mixerutils.h
1104 * added 'sine-tab' waveform to audiotestsrc
1105 * added float audio to audiorate
1107 Bugs fixed since 0.10.1:
1109 * 324216 : [cdparanoia] missing patches from 0.8
1110 * 324696 : [videotestsrc] does not start counting the time from zero...
1111 * 324900 : Problem compiling gst-plugins-base with Forte
1112 * 325984 : [playbin] cannot handle sources that produce raw audio/video
1113 * 325990 : patch videotestsrc for using glib types
1114 * 326601 : GstRingBuffer crashes with alaw/mulaw caps
1115 * 327114 : [theoradec] should post tags on the bus
1116 * 327216 : vorbisdec segfaults on certain queries
1118 API added since 0.10.1:
1120 * added libgstcddabase
1121 * added mixerutils.h
1123 Changes since 0.10.0:
1125 * Parallel installability with 0.8.x series
1126 * Threadsafe design and API
1127 * removed gst-launch-ext
1129 * Fixes for: subparse, xvimagesink, audioresample, videorate, decodebin
1131 Bugs fixed since 0.10.0:
1133 * 322347 : GstBaseRtpDepayload timestamps are wring
1134 * 323900 : Basertpdepayloader lets NEWSEGMENT events through unfiltered
1135 * 323878 : missing < string.h > inclusion (for memset & FD_ZERO)
1137 API added since 0.10.0:
1139 * GstAlsaMixer::device
1140 * GstAlsaMixer::device-name
1142 Bugs fixed since 0.9.7:
1144 * 319172 : gstreamer-plugins-base-0.9.pc doesn't export linking flags
1145 * 323017 : While(1) loop with sleep(0) in basertpdepayload.c
1147 Changes since 0.9.6:
1149 * Parallel installability with 0.8.x series
1150 * Threadsafe design and API
1151 * ximagesink and xvimagesink updates and interactive test
1153 * rename net to netbuffer library
1154 * rtp element renaming
1155 * stream selector fixes
1157 Bugs fixed since 0.9.6:
1159 * 319618 : [decodebin] some ogg videos don't play
1160 * 320644 : RTP packetizer does't set the packet timestamps correctly
1161 * 322388 : xvimagesink force-aspect-ratio=True always displays squar...
1162 * 322704 : oggdemux typefind list leak
1164 Changes since 0.9.5:
1166 * Parallel installability with 0.8.x series
1167 * Threadsafe design and API
1168 * lots of leak fixes
1169 * flicker-free and rewritten X sinks
1170 * fractional framerates
1171 * removed sinesrc, replaced by audiotestsrc
1173 Bugs fixed since 0.9.5:
1175 * 316442 : playbin should use autoaudiosink/autovideosink by default
1176 * 318353 : [ffmpegcolorspace] forward-port fixes from 0.8 branch
1177 * 320200 : vorbisenc: min-bitrate and max-bitrate are 1/1000 bps rat...
1178 * 321164 : gstringbuffer stops working under load
1179 * 321426 : ximage plugin should be renamed to ximagesink
1180 * 321446 : sinesrc should be dropped in favour of audiotestsrc
1181 * 321451 : GstRtpBuffer: no way to create a sub buffer with only the...
1182 * 321816 : [API] xoverlay API to post prepare-xwindow-id message
1183 * 321894 : vorbisenc doesn't compile
1184 * 322117 : Rename libgsttagedit to libgsttag
1186 Changes since 0.9.4:
1188 * video caps now use a good range for framerate and w/h
1189 * oggdemux/oggmux improvements
1190 * playbin improvements
1192 Bugs fixed since 0.9.4:
1194 * 319110 : [PATCH] oggdemux chain finding is slow
1195 * 320058 : playbin of a jpeg over http does not work
1196 * 320923 : [volume] doesn't build on Solaris
1197 * 321011 : gstbasertpdepayload doesn't send the " new segment " event ...
1199 Changes since 0.9.3:
1201 * New element: audiotestsrc
1202 * typefind improvements
1203 * buffer-frames removed
1205 Changes since 0.9.2:
1209 Bugs fixed since 0.9.2:
1211 * 313251 : ximagesink unused functions
1212 * 315159 : audioconvert lost 24 bit conversions in the rewrite
projects / platform / upstream / gstreamer.git / blob