1 This is GStreamer Base Plug-ins 0.10.23, "Emergency de-stress signal"
5 * New navigation API to support DVD playback
6 * playbin2 improvements
7 * RTSP extensions to allow extra headers and options
8 * Replace audioresampler with speexresample based code
9 * Support interlacing flags in the gstvideo library
10 * Support new RIFF formats
12 * Support more frame formats in videoscale
13 * Many other bug-fixes and improvements
15 Bugs fixed since 0.10.22:
17 * 577637 : [playbin2] expose temp-location property
18 * 580120 : [playbin2] unit test fails
19 * 478512 : [alsamixer] volume control slider not working
20 * 574962 : rhythmbox crash in flac_type_find
21 * 564139 : Documentation of TCP plugins
22 * 577436 : xvimagesink should use xcontext- > depth and not count bits...
23 * 350311 : [playbin2] support for subpicture subtitles
24 * 378094 : Enable pango elements to handle UYVY
25 * 543591 : Gnonlin can not play theora streams
26 * 553295 : [riff] fuzzed AVI file causes segfault
27 * 565105 : Gstreamer does not change from READY back to PAUSED in sa...
28 * 565777 : [riff] unrecognised video fourcc 0x10000002 for mpeg2 in avi
29 * 566661 : [typefind] Fall back to file extension using uri query
30 * 567255 : [riff] doesn't detect codec_id 0x706d as AAC (amongst other)
31 * 567636 : [pbutils] Missing plugins code shouldn't ask for the same...
32 * 567740 : bogus warning in decodebin2?
33 * 568482 : linking problems in gst-plugins-base
34 * 569655 : [ffmpegcolorspace] Add UYVY422 to GRAY8 conversion function
35 * 570142 : Documentation is broken for uridecodebin
36 * 570356 : aac typefinder failure
37 * 570768 : [ximagesink] wrong mouse pointer position if output windo...
38 * 570832 : Add flags to enhance mixer interfaces
39 * 571009 : [tagdemux] WMA file with id3v2 tag causes assertion to fail
40 * 571147 : [ffmpegcolorspace/videotestsrc] Add support for packed/pl...
41 * 572577 : [playbin2] deadlock on shutdown
42 * 572872 : [ffmpegcolorspace] Add YVYU colorspace
43 * 572993 : [subparse] broken libregex dependency on Windows
44 * 573165 : Generate additional export files for gstreamer app plugin
45 * 573528 : Wrong format modifier in gstgiobasesink.c
46 * 573529 : In gstrtspconnection.c some functions are called with wro...
47 * 574293 : [decodebin2] deadlock on shutdown
48 * 574319 : Missing HAVE_PROCESS_H in win32/common/config.h
49 * 574447 : gstadder.c: line 904: error C2036: 'gpointer' : unknown size
50 * 574939 : [typefinding] flac typefinder mis-typefinds PDFs as flac ...
51 * 575550 : srt subtitle file keeps playbin2 from playing
52 * 575638 : kissfft copyright
53 * 575649 : [oggdemux] duration query in time format returns true wit...
54 * 576019 : On Windows queue2 can't write files longer than 2-4 GiB, ...
55 * 576142 : [vorbisenc] Non-header output buffers have NULL caps
56 * 576180 : [playbin2] Uses unref'd audiosink volume if using gconfau...
57 * 576586 : [alsamixer] gnome-sound-properties freeze
58 * 577054 : [videoscale] Not valgrind clean
59 * 577709 : Review new navigation API
60 * 577827 : [appsink] Have appsink new_buffer-callback return GstFlow...
61 * 578583 : [PATCH] multifdsink doesn't handle sync-method=latest-key...
62 * 578656 : Implement upstream GstForceKeyUnit events in theoraenc
63 * 579129 : pkgconfig: appsrc/appsink can not be linked to uninstalled
64 * 579130 : app: expose trivial type macros
65 * 579192 : gst_rtcp_packet_get_type should not assert on packet content
66 * 579203 : baseaudiosink: unparenting the ringbuffer in NULL causes ...
67 * 579267 : [rtspconnection] g_async_queue_new_full() is GLib-2.16 AP...
68 * 579463 : [cddabasesrc] [cdparanoiasrc] no longer emits discid
69 * 579668 : audioresample fails to build with --disable-gst-debug
70 * 579734 : [playbin] raw_decoding_mode seems to be set unconditionally
71 * 579912 : [decodebin2] multiqueue is too small in time (interleave ...
72 * 580470 : [audioresample] causes pipelines to go out of sync and be...
73 * 580952 : [audioresample] bad quality/pops compared to plughw
74 * 581727 : [playbin2] make playsink go to PAUSED async
75 * 569682 : playbin2 leaks request pad from input selector
76 * 580020 : [vorbisenc] causes buffers to be out of segment if new se...
77 * 562794 : rtspsrc fails to create a socket on Win32 sometimes.
78 * 567396 : playbin2: DECODE_BIN_LOCK occasionally called twice withi...
79 * 567982 : " queued_bytes " field isn't updated while flushing the que...
80 * 571299 : [appsink] Handoff callback API
81 * 574443 : rtsp win32 - forgotten variable
82 * 574516 : [typefind] add typefinder for photoshop .psd files
83 * 574964 : gst_app_src_end_of_stream(), mutex on error return
84 * 575256 : rtspsrc fails to resolve hostnames
85 * 575588 : decodebin2 deadlock
86 * 576187 : [playbin2] Stalls video sink when disabling subtitles in ...
87 * 576188 : [playbin2] Reusing a playbin2 instance with visualization...
88 * 576190 : [playbin2] Deadlock when reusing playbin2 after an error
89 * 577288 : " Internal playbin error " when seeking to the end of files
90 * 577610 : RTCP feedback messages support in GstRTCPPacket
91 * 577794 : [playbin2] leaks elements set through properties
92 * 578118 : [multifdsink] add option to not resend the streamheader w...
93 * 578506 : Pipeline with alsasrc and alsasink cannot change state ba...
94 * 578942 : Missing RTSP headers related to Windows Media extension.
95 * 580271 : videorate: fails to clear discont flag on duplicated buffers
96 * 580649 : uridecodebin: bug on documentation published in website
98 API added since 0.10.22:
100 * GstRTSP::gst_rtsp_options_as_text()
101 * GstRTSPMessage::gst_rtsp_message_take_header()
102 * GstRTSPRange::gst_rtsp_range_to_string()
103 * New Navigation interface commands, queries and messages
104 * gst_rtsp_channel_new()
105 * gst_rtsp_channel_unref()
106 * gst_rtsp_channel_attach()
107 * gst_rtsp_channel_queue_message()
108 * gst_rtsp_connection_accept()
109 * GstAppSink::gst_app_sink_set_callbacks()
110 * GST_VIDEO_FORMAT_YVYU,GST_VIDEO_BUFFER_TFF,GST_VIDEO_BUFFER_RFF,GST_VIDEO_BUFFER_ONEFIELD
111 * GST_MIXER_FLAG_HAS_WHITELIST,GST_MIXER_FLAG_GROUPING,GST_MIXER_TRACK_NO_RECORD,GST_MIXER_TRACK_NO_MUTE,GST_MIXER_TRACK_WHITELIST
112 * GstAppSrc::emit-signals
113 * GstAppSrc::gst_app_src_set_emit_signals()
114 * GstAppSrc::gst_app_src_get_emit_signals()
115 * GstAppSrc::gst_app_src_set_callbacks()
116 * RTSP::gst_rtsp_connection_get_url()
117 * GstRTSPLowerTrans::GST_RTSP_LOWER_TRANS_HTTP
118 * RTSP:gst_rtsp_connection_set_tunneled()
119 * RTSP:gst_rtsp_connection_is_tunneled()
120 * RTSP::gst_rtsp_connection_set_ip()
121 * RTSP::gst_rtsp_connection_get_tunnelid()
122 * RTSP::gst_rtsp_connection_do_tunnel()
123 * RTSP::gst_rtsp_watch_reset()
127 1) Please note that decodebin2 and playbin2 API included in this release is
128 still considered unstable and WILL change in future releases. At this stage,
129 only developers or early adopters should consider using decodebin2 or playbin2
130 API embodied in their signals and properties.
132 Changes since 0.10.21:
134 * Require gettext 0.17
135 * Replace audioresample with speexresample from -bad
136 * Support new formats in RIFF: uncompressed RGB, WMA lossless, VP6
137 * Move libgstapp and elements from -bad
138 * Support color-key setting and probing for Xv properties
139 * Improve typefinding for various formats
140 * Extend audio sinks for pull-mode operation
141 * Support for more subtitle formats
142 * More development on decode2bin and playbin2
144 * Many bug fixes and improvements
146 Bugs fixed since 0.10.21:
148 * 562163 : theoraenc likely ignoring segments
149 * 562258 : rtspsrc element takes long time to error out if the addre...
150 * 561789 : [volume] deadlocks with a controller attached
151 * 554533 : [xvimagesink] allow setting colorkey if possible
152 * 567511 : colorkey in xvimagesink gets reset when element is reused
153 * 116051 : libresample doesn't handle > factor of 2 rate conversion
154 * 346218 : [audioresample] doesn't do anti aliasing
155 * 385061 : [audioresample?] investigate high CPU usage
156 * 456788 : [subparse] can't handle UTF-16 charset encoded subtitle.
157 * 525807 : [vorbisenc] vorbisenc has problems with a gnlsource that ...
158 * 546955 : gstoggmux EOS handling issue
159 * 549417 : [audioresample] unit test fails on 64bit linux
160 * 549510 : audioresample doesn't negotiate ideal caps
161 * 552237 : UTF-16 srt confuses gstreamer, misdetected as mp3
162 * 552559 : Implementation of SLAVE_SKEW in baseaudiosrc
163 * 552569 : audioresample producing strange sized buffers
164 * 552801 : audioconvert can overflow with big audio buffers
165 * 554879 : Add ability to specify format for date/time display in Gs...
166 * 555257 : Doesn't display srt subtitles saved with BOM
167 * 555319 : add FFV1 fourcc to riff-media
168 * 555607 : subrip subtitles typefind too strict
169 * 555699 : [PATCH] theoradec: prefer container's pixel aspect ratio ...
170 * 556025 : build failure in tests/icles
171 * 556066 : Last byte of FLAC image buffer chopped off
172 * 557365 : subparse check fails
173 * 558124 : [PLUGIN-MOVE] Move speexresample as audioresample2 to -base
174 * 559111 : ALSA sink hangs on USB audio device unplug while playing
175 * 559478 : does not play windows media streams correctly
176 * 559567 : `gst_base_audio_sink_sync_latency' should call `gst_base_...
177 * 561436 : videorate element add image/jpeg to caps template
178 * 561734 : playbin2 additions
179 * 561780 : Playbin2 should work without volume too
180 * 561924 : oggdemux hangs when given corrupt input via non-seekable ...
181 * 562270 : build without gdk fails
182 * 563143 : ximagesink/xvimagesink : _alloc_buffer returns non-clean ...
183 * 563174 : Implement gst_rtcp_packet_remove
184 * 563508 : [rgvolume] Unit test fails with passthrough assertions
185 * 563718 : Theora check out of date
186 * 563904 : GNOME Goal: Clean up GLib and GTK+ includes
188 API added since 0.10.21:
190 * clockoverlay::time-format
191 * GstRingBuffer:gst_ring_buffer_activate()
192 * GstRingBuffer:gst_ring_buffer_is_active()
193 * GstRingBuffer:gst_ring_buffer_convert()
194 * Add GST_TYPE_BASE_AUDIO_(SRC|SINK)_SLAVE_METHOD to the public API
195 * gst_netaddress_get_address_bytes()
196 * gst_netaddress_set_address_bytes()
198 Changes since 0.10.20:
200 * Continue playbin2 development
201 * Ogg improvements - CELT support, skeleton fixes
202 * DVD subpicture support
203 * Improved audio dithering random number generator
204 * xvimagesink/ximagesink fixes
205 * Vorbis encoding and decoding fixes
206 * Recognise Kate subtitle streams
207 * Many bug-fixes and enhancements
209 Bugs fixed since 0.10.20:
211 * 537380 : [gnomevfssrc] Doesn't handle short reads properly
212 * 538656 : xvimagesink support for autofill/colorkey property
213 * 540334 : Build fails without X in tests/examples/seek
214 * 528299 : Multiple GstMixerTracks with the same label cause problem...
215 * 530068 : Ogg Streams with Skeleton and Granulepos > 0 do not work(...
216 * 537009 : playbin2 silly typo breaks signals
217 * 537045 : decodebin2 sometimes emits 'drained' multiple times
218 * 537599 : [oggdemux] skeleton streams not skipped in ogg
219 * 537889 : [xvimagesink] colorbalance is bad
220 * 538232 : vorbisenc/vorbisdec don't work with a live source
221 * 538663 : gdppay memleak in gst_gdp_pay_reset
222 * 540215 : decodebin does not insert a queue for raw data type
223 * 540351 : [avidemux] Doesn't know about Duck DK4 ADPCM
224 * 540497 : ffmpegcolorspace is returning wrong size
225 * 541358 : cross mingw32 gcc: getaddrinfo is not in ws2_32.dll befor...
226 * 544306 : rtspsrc debug=1 segfaults with some libc
227 * 548898 : GStreamer-CRITICAL errors on seeking beyond stream borders
228 * 548913 : vorbisenc being picky about rounding errors in timestamps
229 * 549062 : Video devices aren't updated on subsequent probing.
230 * 549814 : [typefind] add application/pdf typefinder
231 * 550582 : [oggdemux] KATE streams not recognised
232 * 550638 : [typefind] Recognize some jpeg2k file types
233 * 550656 : recognize TrueSpeech in wavparse
234 * 550729 : gst-plugins-base won't compile with " -pedantic " option
235 * 552960 : tagdemux asserts and aborts on truncated files
236 * 553244 : theoraparse doesn't work at all (throws criticals and ass...
238 API added since 0.10.20:
240 * Add "index" property to GstMixerTrack to differantiate between
241 multiple mixer tracks with the same label.
243 Changes since 0.10.19:
246 * Support digest auth for RTSP
247 * Additional documentation
248 * Support DSCP QoS in multifdsink
249 * Add NV12/NV21 video buffer layouts
250 * Video scaling now bilinear by default
251 * Support more than 8 channels in audio conversions
252 * Channel mapping fixes for audioconvert
253 * Improve tmplayer and sami subtitle support
254 * Support 1x1 pixel buffers for videoscale
255 * Typefinding improvements for MPEG2, musepack
256 * Ogg/Dirac mapping updated in oggmux
257 * Fixes in ogg demuxing
258 * audiosink synchronisation and slaving fixes
259 * Support muting of the audio in playbin by selecting -1 as the audio stream
260 * Work done on playbin2 and uridecodebin
261 * Improvements in the experimental GIO plugin
263 * Handle GAP buffers in some places
264 * Various other leak and bug-fixes
266 Bugs fixed since 0.10.20:
268 * 526794 : [giosrc] totem doesn't work with some gvfs backends
269 * 510417 : [PLUGIN-MOVE] Move gio to gst-plugins-base
270 * 509125 : crash in CD Player: - playing CD - lowering/...
271 * 517813 : [audioconvert] make gap aware
272 * 302798 : [playbin] add mute property
273 * 342294 : Setting playbin property current-audio=-1 also stops the ...
274 * 398033 : [audioconvert] support more than 8 channels
275 * 419351 : [avi/a52dec] AV synchronization problems
276 * 467911 : [subparse] sami parser update
277 * 469933 : multifdsink IPv6 and diffserv TOS/TC markup
278 * 506659 : [textoverlay] rendering error when using non-standard widths
279 * 512333 : [gstvorbistag] Retrieve Ogg/Vorbis cover art as image met...
280 * 512382 : [playbin] race condition when pausing/playing multiple in...
281 * 518037 : pbutils-enumtypes.c is not included in win32/vs6/libgstpb...
282 * 521761 : gstaudioclock frozen the clock value until reaches latest...
283 * 522401 : gdpdepay doesn't validate payload CRCs
284 * 523993 : playbin2 blocks after a while when listening to a radio s...
285 * 524724 : [PATCH] [baseaudiosrc] buffer-time and latency-time do no...
286 * 525665 : Crash on Ogg/Vorbis with chain=NULL
287 * 525915 : [streamheader] Unit test fails with " gst_adapter_peek: as...
288 * 526173 : [typefinding] fails to detect mpeg video stream whereas m...
289 * 529018 : gst_ogm_parse_stream_header creates fraction value with w...
290 * 529500 : [videotestsrc] support for NV12 and NV21
291 * 529546 : [Playbin] Memory leak in streaminfo handling
292 * 530068 : Ogg Streams with Skeleton and Granulepos > 0 do not work(...
293 * 530531 : [typefinding] bad read in mpeg_video_stream_type_find
294 * 530719 : gst_video_calculate_display_ratio fails when playing Ogg ...
295 * 530962 : [subparse] parses only every second line of TMPlayer subt...
296 * 532454 : [NV12/NV21] videotestsrc and ffmpegcolorspace don't play ...
297 * 533087 : GstRTSPTransport kept opaque in docs
298 * 533817 : [audioconvert] Can't use default 7 channel layout / only ...
299 * 534071 : Gdppay memleak
300 * 534331 : race in decodebin when changing states while the internal...
301 * 535356 : vorbisdec doesn't support 8 channels
302 * 536475 : gdppay memleak and possible crash
303 * 536521 : Refcounting errors in playbin
304 * 536874 : Build failure on windows
305 * 532166 : [ffmpegcolorspace] support NV12 format
306 * 533617 : [audioconvert] Produces silence when converting 1/2 chann...
307 * 536848 : [giosrc] Doesn't handle short reads properly
308 * 536849 : [giosrc] Very slow doing any playback
309 * 518082 : [alsamixer] playback volumes overwritten by capture volum...
310 * 435633 : [PATCH] videorate not (fully) segment aware; causes frame...
311 * 532364 : tcpclientsrc broken in 0.10.19
312 * 533075 : gst_rtp_buffer_compare_seqnum doesn't do what it says
313 * 533265 : [cddabasesrc] Sound Juicer cut a sector when ripping a track
315 API additions since 0.10.20:
317 * decodebin2::sink-caps property
318 * giosrc::file property
319 * giosink::file property
320 * gst_base_audio_src_set_slave_method()
321 * gst_base_audio_src_get_slave_method()
322 * GstAudioClock::gst_audio_clock_reset()
323 * GstBaseAudioSrc:actual-buffer-time property
324 * GstBaseAudioSrc:actual-latency-time property
325 * gst_audio_check_channel_positions()
326 * add gst_tag_image_data_to_image_buffer()
327 * add gst_tag_list_add_id3_image()
328 * add GST_TAG_IMAGE_TYPE_NONE enum value
330 Changes since 0.10.18:
332 * Handle EAGAIN when polling sockets in rtspconnection
334 Changes since 0.10.17:
336 * Experimental GIO plugin
337 * Continued playbin2 development
339 * Better network element support on Windows
340 * Various other bug-fixes and improvements
342 Bugs fixed since 0.10.17:
344 * 509637 : [API] [basertpaudiopayload] add _set_samplebits_options()
345 * 510229 : [gnomevfssrc] HTTPS support
346 * 511478 : [rtpbuffer] add gst_rtp_buffer_set_extension_data function
347 * 511810 : [RTSP] Uses MT-unsafe gmtime() function
348 * 512899 : [alsa] gstalsasink.c:527: warning: 'snd_pcm_sw_params_set...
349 * 513167 : Fix compiler warning due to disabled signals in mixertrac...
350 * 514307 : [playbin] warning in nautilus, volume element can't be cr...
351 * 514623 : Ogg Theora video slow
352 * 514937 : Correct initialization of hints in is_multicast_address()
353 * 515654 : xvimagesink doesn't build with --disable-xshm
354 * 516246 : [alsasink] handle negative delay from snd_pcm_delay
355 * 517420 : typefind: add h264 elementary stream discovery
356 * 517991 : problems with configure file depending on GCC compiler
357 * 518039 : libgstrtsp MSVC 6.0 compile error
358 * 518162 : [subparse] handle italic text starting with " / " with Micr...
359 * 518940 : [playbin2] make _get_*_tags() match vfuncs prototype in c...
360 * 519906 : [API] add GstMixerOptions::get_values vfunc
361 * 519916 : [API] add mixer-changed and options-list-changed messages
362 * 520523 : [API] Unreviewed changes to ringbuffer API
363 * 521743 : libgstnetbuffer.def exports not up to date
364 * 522625 : [video] gst_video_format_parse_caps() broken for RGBA for...
365 * 523054 : gstbasesrc crashes when called from typefind helpers
366 * 511825 : [RTSP] compiler warning on FreeBSD
367 * 520300 : [alsasrc] provide-clock=false messes up buffer durations
369 API added since 0.10.17:
371 * GstRTPBuffer:gst_rtp_buffer_set_extension_data()
372 * add GST_VIDEO_FORMAT_Y41B and GST_VIDEO_FORMAT_Y42B.
373 * add GstMixerOptions::get_values vfunc (#519906)
374 * add gst_mixer_options_list_changed(), gst_mixer_mixer_changed() and
375 gst_mixer_message_parse_options_list_changed(). Fixes #519916.
376 * gst_base_rtp_audio_payload_set_samplebits_options()
377 * GstNetBuffer::gst_netaddress_equal
379 Changes since 0.10.16:
381 * Work-around ABI breakage due to unfortunate use of the
382 GST_DISABLE_DEPRECATED macro
383 * Export 2 missing functions needed for bindings in the win32 build
384 * Initialise the GstRingBuffer GType from a thread-safe context
386 Bugs fixed since 0.10.16:
388 * 511825 : [RTSP] compiler warning on FreeBSD
389 * 513018 : crash in Volume Control: I typed my password at t...
390 * 512334 : g_critical() when using GstAudioFilter & GST_DEBUG
392 Changes since 0.10.15:
394 * Handle newer Theora granule-pos semantics
395 * Introducing first alpha version playbin2 - the upcoming successor to
397 * Fixes in playbin handling of stream-switching
398 * New API for uniform handling of raw-video format buffers.
399 * Improvements for RTSP/RTP handling
400 * RIFF lib additions for VC-1 and AVC1 fourccs
401 * Many other bug-fixes and improvements
403 Bugs fixed since 0.10.15:
405 * 506132 : Review of changes in video/video.h
406 * 320984 : [oggdemux] cannot handle multiple chains
407 * 373011 : [playbin] throws error when switching off subtitles
408 * 436756 : Intermittent crashes in Pidgin in audioclock g_type_class...
409 * 462740 : [streamselector] patch to improve default stream selection
410 * 486840 : [alsamixer] use _all variants when setting the mixer
411 * 497964 : theoraenc test fails
412 * 498228 : gst-plugins-base-0.10.15 does not compile on FreeBSD (Gen...
413 * 499697 : Provide better pkg-config files
414 * 502497 : [subparse] SubRip subtitles starting from 0 not recognised
415 * 503440 : The control sockets used by gstrtspconnection.c are never...
416 * 503930 : [cdda] warning: 'eos' may be used uninitialized in this f...
417 * 506928 : [alsamixer] add " PCM " as master fall back for cards that ...
418 * 508138 : [decodebin] does not error out if pad activation fails
419 * 509762 : missing file in win32/MANIFEST
420 * 511274 : gst_rtp_buffer_get_extension_data is returning FALSE when...
421 * 496731 : [PATCH] xvimagesink leaks memory if initialization fails
422 * 496761 : [PATCH] RTSP message leaks memory when uninitialized
423 * 500763 : SIGSEGV while playing ogg audio file
425 API additions since 0.10.15:
427 * New GstVideoFormat API and helper functions in libgstvideo
428 * gst_base_audio_sink_set_provide_clock()
429 * gst_base_audio_sink_get_provide_clock()
430 * gst_base_audio_sink_set_slave_method()
431 * gst_base_audio_sink_get_slave_method()
432 * gst_base_audio_src_set_provide_clock()
433 * gst_base_audio_src_get_provide_clock()
435 Changes since 0.10.14:
437 * RTP/RTSP/RTCP/SDP support improved
438 * New FFT support library libgstfft, based on Kiss FFT
439 * New formats supported in volume and audiotestsrc
440 * Fixes in audiorate and videorate
441 * Audio capture fixes
442 * Playbin and decodebin fixes
443 * New tagdemux base class for ID3/APE style tag readers
444 * Fix a nasty crash in the X sinks on shutdown
446 * Add support for multichannel WAV files.
447 * Preserve channel layout information when up/down-mixing.
448 * Many bug-fixes and improvements
450 Bugs fixed since 0.10.14:
452 * 475395 : decodebin2 leaks request-pads
453 * 475451 : [decodebin2] leaks ghostpad
454 * 378770 : [xvimagesink] race condition in event thread?
455 * 407282 : [decodebin2] autoplug-sort signal has GList ** parameter
456 * 430677 : [audioconvert] does not preserve channel positions when f...
457 * 442654 : [volume] controller bypassed by default
458 * 445529 : [volume] support for 24/32-bit audio/x-raw-int
459 * 446766 : return code for gst_base_rtp_payload_audio_handle_event()
460 * 451970 : Subparse requires HTML parser
461 * 453650 : [audiobasesrc] two alsasrcs do not work in one pipeline
462 * 459334 : [textoverlay] expose pango line alignment property
463 * 459585 : [basertpdepayload] api without namespace
464 * 460422 : [audiotestsrc] Add support for float and double output
465 * 462805 : [alsa] compilation fails with gcc 4.2
466 * 462979 : Add 'silent' property to GstTimeOverlay
467 * 463215 : [audioconvert] compile errors
468 * 464320 : [PATCH] gst-plugins-base-0.14 does not build for win32
469 * 464666 : [playbin] QT trailer hangs in preroll with decodebin2
470 * 464690 : Add connection-speed property to uridecodebin element
471 * 465015 : [playbin] Not removed probes causes deadlocks in streamin...
472 * 465028 : some warnings with mingw
473 * 467667 : GST_FRAMES_TO_CLOCK_TIME() and GST_CLOCK_TIME_TO_FRAMES()...
474 * 468129 : [basertpaudiopayload] event handler returns the wrong value
475 * 468619 : New library gstfft: FFT library for integer and float typ...
476 * 470456 : [API] add gst_missing_*_installer_detail_new()
477 * 470766 : [ssaparse] line breaks in SSA subtitle parser
478 * 471067 : Make the SDP code useable for generating SDP descriptions
479 * 471194 : [rtpbuffer] RTP headers are wrong for win32
480 * 473097 : [baseaudiosink] gstreamer-properties hangs when testing s...
481 * 474384 : gstrtsp-enumtypes.c and .h needed for win32
482 * 474880 : [xvimagesink] [ximagesink] leaking buffer caps reference
483 * 475731 : rtspconnection is able to read incomplete messages
484 * 483620 : All Rtp buffers are discarded -- gst_rtp_buffer_get_payl...
485 * 484989 : memleak, not unrefed caps for gstbasertppayload.c
486 * 489010 : Please change default channel order for WAVE_EXT-less .wa...
487 * 491722 : [playbin] regression: crash with external subtitles
488 * 492098 : [GstFFT] Broken scaling
489 * 492114 : Build issues on Windows/MSVC
490 * 492306 : compilation errors with MinGW
491 * 492813 : Missing symbols in libgstrtp.def
492 * 493986 : Build issues on Windows (missing symbols)
493 * 494346 : pre-release vs6 patch
494 * 496548 : Including malloc.h breaks macos build
495 * 496724 : DSW file references non-existent DSP files
496 * 464079 : audiotestsrc doesn't respond to conversion queries properly
497 * 442065 : floatcast.h includes config.h and might break other apps
498 * 466717 : gst_event_new_new_segment_full:assertion `start < = stop' ...
499 * 485753 : Decodebin2 deadlocks when nulling pipeline during typefind
500 * 464028 : Move connection-speed from playbin to playbasebin
502 API added since 0.10.14:
504 * GstTagDemux base class for simple tag demuxers
505 * GstBaseAudioSrc::provide-clock property
506 * gst_rtcp_ntp_to_unix()
507 * gst_rtcp_unix_to_ntp()
508 * gst_rtp_buffer_get_header_len()
509 * gst_rtp_buffer_get_extension_data()
510 * gst_rtp_buffer_compare_seqnum()
511 * gst_rtp_buffer_ext_timestamp()
512 * gst_rtcp_packet_sdes_copy_entry()
513 * gst_install_plugins_supported()
514 * gst_missing_*_installer_detail_new() convenience API
515 * gst_rtsp_connection_poll()
516 * GstTextOverlay::line-alignment property
518 Changes since 0.10.13:
520 * Audio dither and noise-shaping when reducing bit-depth
521 * RTSP and SDP helper libraries added
522 * Experimental buffering element "queue2" now supports pull-mode
523 and file-based buffering.
524 * Support for more 32-bit video pixel layouts
525 * Various fixes and improvements
527 Bugs fixed since 0.10.13:
529 * 380625 : [x*imagesink] add 'handle-expose' property
530 * 385527 : oggmux sometimes gets DELTA flag on output wrong near start
531 * 402076 : videoscale 4-tap method broken for downscaling
532 * 437169 : [xvimagesink] add property to disable Xv double-buffering
533 * 441264 : queue2 support to do buffering on a file
534 * 442553 : [v4lsrc] doesn't output segments in GST_FORMAT_TIME
535 * 442557 : [videorate] doesn't handle latency queries
536 * 442944 : Audiotestsrc can overflow on seeks
537 * 444523 : [queue2] Pull mode support
538 * 444630 : Compilation error with fsseko (from gstqueue2.c) -- unabl...
539 * 445505 : [queue2] It does not work in pull mode with oggdemux
540 * 446551 : [queue2] Buffering is not working properly if it is set t...
541 * 446572 : [queue2] Division by zero
542 * 446972 : warning when compiling gstoggdemux.c
543 * 449156 : Regression in CVS for decodebin2
544 * 450875 : Missing files in po/POTFILES.in
545 * 451707 : [tag] UTF-8 in ID3v1 tag not correctly decoded
546 * 451908 : [ffmpegcolorspace] regression: doesn't accept GST_VIDEO_C...
547 * 454264 : Playbin fails to " play " image url after a movie url
548 * 456656 : [API] Addition of audio buffer clipping function to gstaudio
549 * 460978 : gst_audio_buffer_clip outputs warnings
550 * 152864 : [PATCH] GstAlsaMixer doesn't support signals
551 * 360246 : [audioconvert] Optionally apply dithering
552 * 394061 : Add support for Subviewer subtitles
553 * 420326 : Base payloader class has wrong property types and ranges
554 * 451145 : [vorbisdec] errors out on 0-sized packets
555 * 459204 : [PATCH] [playbin] gst_play_base_bin_get_streaminfo_value_...
557 API added since 0.10.13:
559 * RTSP and SDP libraries added
560 * gst_rtsp_base64_decode_ip
561 * Add buffer clipping function gst_audio_buffer_clip for raw audio
562 buffers. Fixes #456656.
563 * gst_mixer_get_mixer_flags
564 * gst_mixer_message_parse_mute_toggled
565 * gst_mixer_message_parse_record_toggled
566 * gst_mixer_message_parse_volume_changed
567 * gst_mixer_message_parse_option_changed
568 * GstMixerMessageType
571 Changes since 0.10.12:
572 * Many fixes and improvements
573 * RTP and RTCP support improved
575 Bugs fixed since 0.10.12:
577 * 339838 : [audioconvert] support floats with non-native endianness
578 * 393975 : closing x/xvimagesink window crashes gst-launch
579 * 405072 : [API] add gst_tag_freeform_string_to_utf8()
580 * 413799 : [subparse] add support for MPL2 format
581 * 414645 : GstMixerTrack should make untranslated label available
582 * 420079 : [audioconvert] Uses biased rounding which results in dist...
583 * 420578 : [subparse] add more colour map in sami parser
584 * 421834 : videorate breaks on dimension changes
585 * 423051 : Vorbis tags of type double use locale-dependent formatting
586 * 423055 : Verify ReplayGain vorbistag processing in libs/tag testsuite
587 * 425455 : Decodebin2 leaks pads
588 * 426250 : GstPlayBaseBin leaks streaminfo objects
589 * 428187 : Rtp base depayloader class doesn't send new_segment after...
590 * 431672 : gst_base_rtp_audio_payload_push() should take object of i...
591 * 432362 : [ximagesink] doesn't build if XShm is not available
592 * 432755 : [videorate] leaks buffer if flow != OK
593 * 432984 : [baseaudiosrc] misleading warning message when dropping s...
594 * 433888 : [theoradec] does not generate a perfect stream
595 * 436562 : Theoradec doesn't work well with gnonlin
596 * 438840 : [theoradec] does not compile with old version of libtheora
597 * 440997 : [gstriff] Doesn't handle width!=depth files with audio/x-...
598 * 441295 : audioconvert doesn't build on VS6
599 * 442024 : regression in playbin buffering
600 * 350299 : [playbin] " Internal data flow error " opening movie with s...
601 * 410039 : totem crashed with SIGSEGV in new_decoded_pad_full()
602 * 340842 : do latency calculation for live sources
603 * 341078 : RB does not play beyond initially downloaded podcast file
604 * 414496 : [id3demux, id3v2mux] Add support for GST_TAG_MUSICBRAINZ_...
606 API additions since 0.10.12:
608 * add gst_tag_freeform_string_to_utf8()
609 * GstRTPBuffer::gst_rtp_buffer_default_clock_rate()
610 * GstBaseAudioSink::slave-method property
611 * add "min-ptime" property to RTP base audio payloader
612 * gst_base_rtp_audio_payload_push()
613 * gst_base_rtp_audio_payload_get_adapter()
614 * GstMixerTrack::untranslated-label property
616 Changes since 0.10.11:
618 * New API for on-demand plugin installation
619 * Xv thread-safety and configuration enhancements
620 * decodebin2 improvements
621 * Support more raw audio format conversions
622 * Improvements in Ogg support
623 * AudioFilter base class ported to 0.10
624 * Fixes for subtitles
625 * Latency/live-playback support for Alsa
626 * Lots of bug fixes and improvements
628 Bugs fixed since 0.10.11:
630 * 398721 : No video in .ogm files with decodebin2
631 * 339837 : [audioconvert] support for 64-bit float audio
632 * 341524 : [decodebin] can't handle decoders with always src pads wi...
633 * 352069 : Add de.po German translation
634 * 363379 : [oggmux] doesn't detect EOS on all sinkpads
635 * 378436 : [oggdemux] rhythmbox crash on fast clicking on rating in ...
636 * 380342 : Totem does not play mp3 files when lyrics are present
637 * 383195 : [cddabasesrc,basertpaudiopayload] compile errors with gcc...
638 * 383198 : totem crashed to gst_xvimagesink_update_colorbalance
639 * 384008 : [xvimagesink] accesses - > xwindow outside locks
640 * 384060 : gst_xoverlay_set_xwindow_id() causing lockups with x(v)im...
641 * 387138 : x input events processing in sinks with xoverlay interfac...
642 * 390063 : Documentation typo
643 * 390076 : add xv adaptor and port properties in xvimagesink element.
644 * 391365 : [oggdemux] internal stream error on OggFlac
645 * 392070 : [vorbis] GST_TAG_LOCATION not mapped
646 * 392393 : [API] add libgstbaseutils library for missing plugins mes...
647 * 396042 : mpeg4 video typefinder loops endlessly on quicktime redirect
648 * 396835 : audioconvert/audioresample combination causing buffer of ...
649 * 397673 : [patch] XIOError caught in x[v]imagesink.c
650 * 397810 : [typefinding] .vob file: could not determine type of stream
651 * 398110 : [theoraenc] GLib failed to allocate 3080991032 bytes on g...
652 * 399340 : Crash in the oggdemux plugin when trying to play a specia...
653 * 401029 : [playbin] rapidly changing visualisation freezes
654 * 401072 : Move libgimme-codec helper functions to GStreamer
655 * 402505 : visualisations don't work for some samplerates
656 * 407811 : decodebin2 hang on HD clip
657 * 409683 : Crash with Decodebin2
658 * 410396 : not reading " DATE " tags from Flac files
659 * 410963 : Fails to build with -z defs
660 * 357503 : [suparse] wrong timing with microdvd subtitles
661 * 393310 : [pango] localtime_r does not exist in MinGW
662 * 397207 : Test failure w/ HP-UX 11.11 & native compiler
663 * 399948 : [textoverlay] leaks upstream events if textpad unlinked
664 * 403963 : GstAudioFilter base class broken
665 * 404512 : [videoscale] floating point exception on 1x1 video
666 * 405020 : [alsa] probing the device-name doesn't seem to work corre...
667 * 408278 : [videorate] memory leak
668 * 410772 : Crash copying a GstNetBuffer
669 * 401118 : [visual] error if width not a multiple of 4
670 * 405451 : [alsasink] deadlocks when disconnecting USB Sounddevice
672 API additions since 0.10.11:
675 * GST_VIDEO_SINK_CAST()
676 * gst_pb_utils_add_codec_description_to_tag_list()
677 * gst_pb_utils_get_codec_description()
678 * gst_pb_utils_get_source_description()
679 * gst_pb_utils_get_sink_description()
680 * gst_pb_utils_get_decoder_description()
681 * gst_pb_utils_get_encoder_description()
682 * gst_pb_utils_get_element_description()
683 * gst_pb_utils_init()
684 * gst_install_plugins_context_new()
685 * gst_install_plugins_context_set_xid()
686 * gst_install_plugins_context_free()
687 * gst_install_plugins_async()
688 * gst_install_plugins_sync()
689 * gst_install_plugins_return_get_name()
690 * gst_install_plugins_installation_in_progress()
691 * gst_missing_uri_source_message_new()
692 * gst_missing_uri_sink_message_new
693 * gst_missing_element_message_new
694 * gst_missing_decoder_message_new
695 * gst_missing_encoder_message_new
696 * gst_missing_plugin_message_get_installer_detail
697 * gst_missing_plugin_message_get_description
698 * gst_is_missing_plugin_message
700 Bugs fixed since 0.10.10:
702 * 360552 : [riff] [avi] extracts non-UTF8 metadata
703 * 365501 : [x/xvimagesink] race condition when creating first image ...
704 * 339366 : [playbin] hangs if suburi file type cannot be determined
705 * 355914 : libvisual causes xvimagesink: assertion `GST_CAPS_REFCOU...
706 * 363118 : gst_riff_create_video_caps() should also store variant in...
707 * 363607 : xvimagesink xwindow_draw_border() slowness
708 * 336301 : [playbin] can't handle RTSP source
709 * 337026 : oggmux doesn't set EOS properly
710 * 337031 : vorbisdec outputs too much data
711 * 340049 : New BaseRTPAudioPayloader class to -base
712 * 348264 : Theora encoding, Ogg muxing don't handle discontinuities
713 * 354773 : xvimage assumes that XV_COLORKEY can be set in RGB888 format
714 * 355917 : libvisual plugin is broken
715 * 355935 : multifdsink doesn't allow setting maximums (soft, hard) i...
716 * 357038 : [ffmpegcolorspace] RGBA handling broken
717 * 357215 : [playbin] buffering notification not quite right yet
718 * 357289 : [riff] riff parser can't detect aac audio stream
719 * 357404 : [playbin] Linking can fail silently
720 * 357531 : [subparse] problem if markup is not closed
721 * 357577 : [playbin] regression: buffering still images broken
722 * 357591 : Avoid compiler warning with uclibc and -Werror
723 * 357613 : XvStopVideo in xvimagesink
724 * 357800 : [libvisual] doesn't pass audio data to libvisual 0.4.0 co...
725 * 359580 : tcpserversink and dataprotocol assert for multipart streams
726 * 361095 : Fixes compiling with forte: warning clean up (part 3)
727 * 361456 : [basertppayload] Memory leak
728 * 361634 : sink- > ringbuffer NULL in BaseAudioSink's setcaps()
729 * 361984 : [subparse] doesn't accept .srt file that doesn't start wi...
730 * 366334 : [PATCH] Windows vs8 fixes
731 * 368273 : Using the remove signal on multifdsink is not threadsafe
732 * 368310 : include file gstbasertpaudiopayload.h not included for r...
733 * 369482 : [typefind] MPEG system streams get recognized as mp3 files
734 * 370092 : [PATCH] Decodebin v2 : Implementation
735 * 377183 : regression: no eos when playing ogg vorbis files
736 * 381219 : bad debugging code left in audiorate
737 * 382223 : [decodebin] more delayed linking
738 * 382269 : Typefind detects mpeg video clip as audio/mpeg
739 * 335635 : Add an Ogg/Vorbis retagging element
740 * 341681 : [textoverlay] flickering with continuously timestamped text
741 * 342228 : [alsa] Recognize " Front " as a Master channel
742 * 357330 : [subparse] some sami parser minor but enhanced patch
743 * 357532 : [gsttag] vorbistag doesn't handle dates that include time...
744 * 359237 : [typefinding] doesn't recognize XML files shorter than 25...
745 * 362845 : [subparse] add support for tmplayer format
746 * 357977 : [videorate] new segment start is not respected
747 * 364812 : [PATCH] oggmux release pad does not remove pad
748 * 364856 : pngenc stride problems
749 * 372507 : Mac build fixes
751 API added since 0.10.10:
753 * playbin::queue-min-threshold property.
754 * GstVideoOrientation interface
755 * gst_base_rtp_depayload_push_ts
756 * gst_base_rtp_depayload_push
757 * Add dropped_buffers to multifdsink's get-stats GValueArray
758 * gst_ring_buffer_commit_full
760 Changes since 0.10.9:
762 * New elements: gdppay, gdpdepay
764 Bugs fixed since 0.10.9:
766 * 343787 : The adder cannot handle when multiple elements tries to l...
767 * 336075 : ALSA emu10k1 mixer tracks are wrongly classified as playb...
768 * 349105 : crash with playbin and resizing screen
769 * 342494 : [v4l] Query " device-name " even if device is not open
770 * 342680 : [adder] seeking with multiple ogg files fails to work
771 * 345188 : [alsa] can't handle more than 8 channels
772 * 347091 : converting vorbis comments to GstTagLists is lossy
773 * 348157 : Changed " Change Device " menu behaviour in gnome-volume-co...
774 * 348916 : [typefind] add multipart/x-mixed-replace typefinder
775 * 350157 : [riff] riff parser can't detect dts audio stream
776 * 350655 : [oggdemux] should process seeking queries
777 * 350900 : [adder] should not clamp floating point values
778 * 351426 : API: add gst_tag_parse_extended_comment
779 * 351502 : g_value_set_string leaks
780 * 351742 : [vorbisenc] discontinuity detection too sensitive, might ...
781 * 353658 : [videotestsrc] doesn't round strides correctly for YVYU
782 * 354594 : multifdsink doesn't work reliably with sync-method = 'nex...
783 * 351790 : [ogmparse] crash parsing video stream on x86-64
784 * 140139 : [avidemux] can't play broken avi with ogg (not vorbis) au...
785 * 347783 : [PLUGIN-MOVE] GDP elements should be moved
786 * 347918 : Internal data flow error in udpsrc
787 * 349656 : jitterbuffer in GstBaseRtp fails to handle rtp seqnum rol...
788 * 350784 : element alsamixer doesn't respect asoundrc
789 * 351308 : [netbuffer] build fails with gkt-doc critical warnings
790 * 353234 : audiorate preserves DISCONT on buffers
791 * 353912 : Add cmml caps to oggmux
793 API added since 0.10.9:
795 * gst_rtp_buffer_get_payload_subbuffer()
796 * gst_tag_parse_extended_comment()
797 * GstPlayBin::connection-speed
798 * GstTheoraParse::synchronization-points
799 * GST_AUDIO_CHANNEL_POSITION_NONE
801 Changes since 0.10.8:
803 * Parallel installability with 0.8.x series
804 * Threadsafe design and API
806 * Support for images in tags
807 * Playback improvements
808 * Gnomevfssrc now supports burn:// uris
809 * Videoscale now supports more RGBA formats
810 * Multifdsink improvements
811 * Testsuite can now generate coverage information
813 Bugs fixed since 0.10.8:
815 * 347296 : Problems with clocks on alsasrc hangs the application
816 * 347295 : [vorbisdec] Pushes before being initialized
817 * 329798 : [playbin] doesn't always give correct error message for m...
818 * 342085 : [alsasink] doesn't set buffer-time correctly
819 * 342789 : [audioresample] doesn't clear state when stopped, causing...
820 * 343303 : [subparse] workaround for bad entities in sami parser
821 * 343385 : [gnomevfs] add support for burn:// URIs
822 * 343500 : [riff] gst_riff_parse_strf_vids() can't parse extra data.
823 * 343699 : oggmux leaks
824 * 344503 : [subparse] parse font face property in sami parser.
825 * 345131 : [PATCH] videoscale support for 32-bit RGB-formats
826 * 345206 : [textoverlay] crash with non-UTF8 input
827 * 345225 : [theoradec] Clipping for exact seeking
828 * 345641 : [API] [libgsttag] add enums for image tag type
829 * 345879 : [riff] won't play a .wmv file with WMVA video stream
830 * 346581 : [typefinding] recognise text/html
831 * 347221 : [audioconvert] channel remapping does not work right
832 * 347304 : Massive leaks with xvimagesink
833 * 346527 : alsasrc get_range does not respect requested size
835 Changes since 0.10.7:
837 * alsasink probing fixes
838 * xvimagesink error reporting fixes
841 * vorbis multichannel fixes
842 * multifdsink streamheader fixes
844 Bugs fixed since 0.10.7:
846 * 169936 : [subparse] support for SAMI subtitles
847 * 315312 : Gstreamer Xv uses RGB instead of YUV.
848 * 334002 : video4linux shouldn't depend on X in configure script
849 * 336881 : [libvisual] additional support for libvisual-0.4
850 * 337544 : [xvimagesink] Internal Error when image is too large
851 * 339520 : [subparse] add " encoding " property
852 * 340909 : [alsasink] can't enable spdif output
853 * 341542 : some users have an assertion failed: (GST_VIDEO_SINK_WIDT...
854 * 341562 : audioconvert doesn't list formats in order of preference
855 * 341696 : audioconvert crashes if converting from a format with no ...
856 * 341719 : bisection algorithm in ogg doesn't bisect in some cases
857 * 341732 : [alsasink] doesn't query supported sample rates
858 * 341873 : [alsasink] minor memory leak, uses unprotected static var...
859 * 342143 : [subparse] sami parser needs to escape characters
860 * 342181 : [alsa] add property probe interface to alsasink and alsasrc
861 * 342268 : [playbin] add 'subtitle-encoding' property
862 * 342345 : [riff] Elephant's Dream AVI does not play, JUNK chunk bef...
863 * 342566 : Building without GTK+ fails
864 * 343397 : H.264/AAC movie deadlocks with totem in gstreamer code, p...
865 * 339935 : [adder] dead-locks when adding sink pads in PAUSED state
867 Changes since 0.10.6:
869 * typefind improvements
870 * bug-fixes in textoverlay, audioconvert, videotestsrc,
871 multifdsink and audio source/sink base classes
872 * Ice-cast metadata support has moved from gnomevfssrc to the
873 icydemux element in gst-plugins-good
874 * audioresample now supports floating point samples
875 * Adder element fixes.
876 * Fixes for network playback and audio resampling in playbin
878 Bugs fixed since 0.10.6:
880 * 340060 : [adder] handle newsegment events properly
881 * 340375 : [API 0.11] [patch] typefind to differentiate between mp4 ...
882 * 339405 : [textoverlay] can't display '\n' character
883 * 338657 : [patch] adder should send events from src-pad to all sink...
884 * 338919 : [patch] alsasink should also query witdh capabilities fro...
885 * 301759 : [audioresample] float audio support (for OSX audio sinks)
886 * 331901 : [videotestsrc] framerate=0/1 gives assertion error
887 * 333657 : Replacing icy demuxing in gnomevfssrc
888 * 336339 : [audioresample] should support width != 16
889 * 338718 : [patch] [audioconvert] correctly clip float samples > 1.0
890 * 338778 : [patch] Bad audio with ASX files
891 * 338991 : [patch] Videoscale doesn't pass on pixel-aspect ratio
892 * 339574 : [patch] Race condition in multifdsink can lead to spuriou...
893 * 339786 : [typefinding] wavpack typefinding doesn't always work
894 * 340369 : [volume element] " volume " property range insufficient
895 * 340379 : [playbin] doesn't insert audioresample, causes problems w...
896 * 340392 : Problem with internal-decodebin
897 * 341160 : [multifdsink] client_status enum has an uninitialized nick
898 * 341182 : Accessing playbin's streaminfo property from high languag...
899 * 341432 : [playbin] automatically get icecast metadata requiring ic...
900 * 341542 : some users have an assertion failed: (GST_VIDEO_SINK_WIDT...
901 * 341557 : Map GST_TAG_IMAGE < = > ID3v2 APIC tag
903 API added since 0.10.6:
905 * client-fd-removed signal added to multifdsink
906 * stream-info-value-array property added to playbin
907 * gst_video_calculate_display_ratio() in libgstvideo
909 Changes since 0.10.5:
911 * QoS in sinks and transform elements
912 * Needs GStreamer 0.10.5 for new GstBaseSink::async_playback() vmethod
913 * added theoraparse element
915 Bugs fixed since 0.10.5:
917 * 313136 : [playbin] hang while playing truncated ogg file
918 * 172848 : [subparse] subtitles with special chars are displayed as ...
919 * 305279 : [riff] uncompressed AVIs with 24bpp don't work
920 * 320765 : [ffmpegcolorspace] make win32+msvc compliant, don't use _...
921 * 323852 : Disable tests/icles on platforms that do not have X
922 * 325653 : build errors compiling audioresample on win32(vs7)
923 * 327357 : gst-plugins-base fails to compile with GCC 4.1
924 * 334620 : [gnomevfssrc] fails to connect to icecast streaming servers
925 * 334822 : [ffmpegcolorspace] YVU9 support
926 * 335028 : [typefinding] ID3 v1 tag is not recognized with mp3-in-wa...
927 * 335365 : inefficient use of GList in gst-plugins-base
928 * 336190 : [gnomevfssink] should accept non-URI filenames as " location "
929 * 336194 : [gnomevfssrc] some minor memory leaks
930 * 336477 : plugins need better/univied descriptions
931 * 336617 : Unable to recognise MPEG TS stream
932 * 337548 : Memory leaks in basertpdepayload
933 * 337945 : [oggdemux] segment stop position ignored
934 * 338419 : Regression in the handling of files with multiple audio/s...
935 * 338897 : Videoscale crashes as part of DVD to Ogg transcoding
936 * 339013 : [videorate] Goes into an infinite loop
937 * 339047 : [riff] handle H264 fourcc in addition to h264
938 * 339212 : ISO file typefinding regression
939 * 330748 : deadlock in base audio sink on playing- > paused state change
941 Bugs fixed since 0.10.4:
943 * 334216 : [gnomevfssrc] won't open some media on NFS mounts any longer
944 * 334226 : typefindfunctions plugin crashes on PPC on registration
946 Changes since 0.10.3:
948 * (Experimental) QoS support
949 * oggmuxer now creates 100% valid streams for Theora, Vorbis and Speex
950 * documentation updates
951 * better support for subtitles (seeking)
953 Bugs fixed since 0.10.3:
955 * 310202 : [subtitles] < i > < /i > tags and others should be supported i...
956 * 312439 : XVideo output doesn't work on remote displays (probably r...
957 * 321271 : audio output is truncated at EOS
958 * 321650 : Can't decode this ogm file
959 * 325732 : [oggdemux] problem when seeking to time less than 4s with...
960 * 325972 : [typefinding] doesn't recognise this mp3
961 * 326720 : [alsasink] doesn't support more than 2 channels anymore
962 * 330711 : [ffmpegcolorspace] problems with palettized RGB (fencount...
963 * 330789 : gstbaseaudiosink causes noise on seeking
964 * 330888 : Fix build with gcc 2.95 (again)
965 * 331295 : gnomevfssink doesn't respect umask when creating files
966 * 331526 : 3GP type detection is too simple
967 * 331678 : Decodebin is not reusable within a single pipeline (as in...
968 * 331690 : playbin won't play my last.fm stream
969 * 331763 : [alsamixer] unmute sets the volume to 100%
970 * 331765 : [alsamixer] mixer applet slider doesn't want to move from...
971 * 331903 : [videorate] doesnt handle input caps of framerate=0/1 sanely
972 * 332778 : [ogmparse] " Already an existing pad " WARNING
973 * 332964 : random crashes in mp3_type_find
974 * 333254 : theora encoder does not set IN_CAPS flag properly
975 * 333352 : [gnomevfssink] reports disk full as generic error
976 * 333488 : Allow for palette < 256 colours in AVI files
977 * 333510 : [PATCH] Fix gst_pad_new_from_template (gst_static_pad_tem...
978 * 333545 : [riff] set depth on wma caps to make asfdemux and pitfdll...
979 * 333663 : [patch] unref the result of gst_pad_get_parent
980 * 333900 : [typefind] cannot play a particular mp3 file
981 * 334112 : variable not initialized
982 * 334129 : Disable frame dropping for now
983 * 317038 : use default channel layout if none is specified in multic...
984 * 319340 : [cdparanoia] uncorrected-error signal never fired
986 API added since 0.10.3:
988 * GstTextOverlay::halignment
989 * GstTextOverlay::valignment
991 Changes since 0.10.2:
993 * typefind improvements
994 * Ogg decoding and encoding fixes
995 * Improved audio and video sink classes
997 * Improved video scaling
998 * On-the-fly visualisation switching
1001 Bugs fixed since 0.10.2:
1003 * 330244 : gsttextoverlay.c:895: 'struct _GstCollectData' has no mem...
1004 * 324000 : [playbin] post error or message on unknown input
1005 * 153004 : [typefind] can't identify mp3 file with one single mpeg f...
1006 * 323874 : [playbin] leaks sinks and threads when using gconfaudiosink
1007 * 324626 : ffmpegcolorspace support for fourcc " UYVY "
1008 * 326447 : check that all elements in -base pass queries they can't ...
1009 * 328263 : Fix build with gcc 2.95
1010 * 328279 : [decodebin] timeout issue when pre-rolling
1011 * 329326 : Fix oggmux removing pads from collect pads
1013 Changes since 0.10.1:
1015 * ported gnomevfssink, cdparanoia
1016 * New library and base class: GstCddaBaseSrc
1017 * ported mixerutils.h
1018 * added 'sine-tab' waveform to audiotestsrc
1019 * added float audio to audiorate
1021 Bugs fixed since 0.10.1:
1023 * 324216 : [cdparanoia] missing patches from 0.8
1024 * 324696 : [videotestsrc] does not start counting the time from zero...
1025 * 324900 : Problem compiling gst-plugins-base with Forte
1026 * 325984 : [playbin] cannot handle sources that produce raw audio/video
1027 * 325990 : patch videotestsrc for using glib types
1028 * 326601 : GstRingBuffer crashes with alaw/mulaw caps
1029 * 327114 : [theoradec] should post tags on the bus
1030 * 327216 : vorbisdec segfaults on certain queries
1032 API added since 0.10.1:
1034 * added libgstcddabase
1035 * added mixerutils.h
1037 Changes since 0.10.0:
1039 * Parallel installability with 0.8.x series
1040 * Threadsafe design and API
1041 * removed gst-launch-ext
1043 * Fixes for: subparse, xvimagesink, audioresample, videorate, decodebin
1045 Bugs fixed since 0.10.0:
1047 * 322347 : GstBaseRtpDepayload timestamps are wring
1048 * 323900 : Basertpdepayloader lets NEWSEGMENT events through unfiltered
1049 * 323878 : missing < string.h > inclusion (for memset & FD_ZERO)
1051 API added since 0.10.0:
1053 * GstAlsaMixer::device
1054 * GstAlsaMixer::device-name
1056 Bugs fixed since 0.9.7:
1058 * 319172 : gstreamer-plugins-base-0.9.pc doesn't export linking flags
1059 * 323017 : While(1) loop with sleep(0) in basertpdepayload.c
1061 Changes since 0.9.6:
1063 * Parallel installability with 0.8.x series
1064 * Threadsafe design and API
1065 * ximagesink and xvimagesink updates and interactive test
1067 * rename net to netbuffer library
1068 * rtp element renaming
1069 * stream selector fixes
1071 Bugs fixed since 0.9.6:
1073 * 319618 : [decodebin] some ogg videos don't play
1074 * 320644 : RTP packetizer does't set the packet timestamps correctly
1075 * 322388 : xvimagesink force-aspect-ratio=True always displays squar...
1076 * 322704 : oggdemux typefind list leak
1078 Changes since 0.9.5:
1080 * Parallel installability with 0.8.x series
1081 * Threadsafe design and API
1082 * lots of leak fixes
1083 * flicker-free and rewritten X sinks
1084 * fractional framerates
1085 * removed sinesrc, replaced by audiotestsrc
1087 Bugs fixed since 0.9.5:
1089 * 316442 : playbin should use autoaudiosink/autovideosink by default
1090 * 318353 : [ffmpegcolorspace] forward-port fixes from 0.8 branch
1091 * 320200 : vorbisenc: min-bitrate and max-bitrate are 1/1000 bps rat...
1092 * 321164 : gstringbuffer stops working under load
1093 * 321426 : ximage plugin should be renamed to ximagesink
1094 * 321446 : sinesrc should be dropped in favour of audiotestsrc
1095 * 321451 : GstRtpBuffer: no way to create a sub buffer with only the...
1096 * 321816 : [API] xoverlay API to post prepare-xwindow-id message
1097 * 321894 : vorbisenc doesn't compile
1098 * 322117 : Rename libgsttagedit to libgsttag
1100 Changes since 0.9.4:
1102 * video caps now use a good range for framerate and w/h
1103 * oggdemux/oggmux improvements
1104 * playbin improvements
1106 Bugs fixed since 0.9.4:
1108 * 319110 : [PATCH] oggdemux chain finding is slow
1109 * 320058 : playbin of a jpeg over http does not work
1110 * 320923 : [volume] doesn't build on Solaris
1111 * 321011 : gstbasertpdepayload doesn't send the " new segment " event ...
1113 Changes since 0.9.3:
1115 * New element: audiotestsrc
1116 * typefind improvements
1117 * buffer-frames removed
1119 Changes since 0.9.2:
1123 Bugs fixed since 0.9.2:
1125 * 313251 : ximagesink unused functions
1126 * 315159 : audioconvert lost 24 bit conversions in the rewrite
projects / platform / upstream / gstreamer.git / blob