1 2009-10-01 10:37:38 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3 * ext/pango/gsttextoverlay.c:
4 * ext/pango/gsttextrender.c:
5 pango: Unpremultiply Cairo's ARGB to match GStreamers ARGB
7 2009-09-28 22:06:11 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
9 * gst/playback/gstplaysink.c:
10 playsink: make the lock recursive for now
13 2009-09-28 21:54:03 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
15 * gst/playback/gstplaysink.c:
16 playsink: fix the vis property getter
18 2009-09-30 18:06:56 +0100 Christian F.K. Schaller <christian.schaller@collabora.co.uk>
20 * gst-plugins-base.spec.in:
21 Add missing file to spec file
23 2009-09-17 16:57:48 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
25 * gst-libs/gst/cdda/gstcddabasesrc.c:
26 * tests/check/libs/cddabasesrc.c:
27 cddabasesrc: Fix string leaks in the unit test and a leak in cddabasesrc
29 2009-09-17 23:42:52 +1000 Jonathan Matthew <jonathan@d14n.org>
31 * gst-libs/gst/cdda/gstcddabasesrc.c:
32 * tests/check/libs/cddabasesrc.c:
33 cddabasesrc: ignore URI fragments that look like device paths
34 Rhythmbox uses cdda:// URIs of the form cdda://track#device, which
35 worked before the fix for bug #321532.
36 Also adds a check for negative track numbers and some unit tests for URI
40 2009-09-17 01:20:45 +0100 Jan Schmidt <thaytan@noraisin.net>
77 2009-09-15 15:23:49 -0700 Michael Smith <msmith@songbirdnest.com>
79 * gst-libs/gst/tag/gstvorbistag.c:
80 vorbistag: don't ever return NULL in list of strings.
82 2009-09-14 12:18:33 +0200 Edward Hervey <bilboed@bilboed.com>
84 * gst/playback/gstplaysink.c:
85 playsink: Expose mute,volume,vis-plugin and font-desc properties
86 https://bugzilla.gnome.org/show_bug.cgi?id=594623
88 2009-09-09 12:42:04 +0200 Edward Hervey <bilboed@bilboed.com>
90 * gst/playback/gstplaysink.c:
91 GstPlaySink: Expose 'reconfigure' as an action signal.
93 2009-09-09 11:17:28 +0200 Edward Hervey <bilboed@bilboed.com>
95 * gst/playback/gstplaysink.c:
96 GstPlaySink: Expose flags as a gobject property.
98 2009-09-08 11:35:20 +0200 Edward Hervey <bilboed@bilboed.com>
100 * gst/playback/gstplayback.c:
101 * gst/playback/gstplaysink.c:
102 * gst/playback/gstplaysink.h:
103 playback: Register playsink as an element.
104 This allows using playsink from outside the playback plugin.
105 Add code to be able to request the sink pads using standard GStreamer API.
106 TODO : expose GObject properties/signals.
108 2009-09-12 14:55:06 +0300 Stefan Kost <ensonic@users.sf.net>
110 * docs/libs/gst-plugins-base-libs.types:
111 docs: add new gst_stream_volume_get_type to types file
112 This is needs to get Gobject features to show up in the docs.
114 2009-09-12 15:48:11 -0700 David Schleef <ds@schleef.org>
116 * ext/ogg/gstoggdemux.c:
117 oggdemux: Fix duration calculation for truncated files
118 If the last page of a stream has a granulepos of -1, that is,
119 it doesn't complete a packet, we need to continue to search
120 for the last granulepos.
122 2009-09-12 14:01:20 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
125 * gst-libs/gst/app/Makefile.am:
126 * gst-libs/gst/audio/Makefile.am:
127 * gst-libs/gst/cdda/Makefile.am:
128 * gst-libs/gst/fft/Makefile.am:
129 * gst-libs/gst/interfaces/Makefile.am:
130 * gst-libs/gst/netbuffer/Makefile.am:
131 * gst-libs/gst/pbutils/Makefile.am:
132 * gst-libs/gst/riff/Makefile.am:
133 * gst-libs/gst/rtp/Makefile.am:
134 * gst-libs/gst/rtsp/Makefile.am:
135 * gst-libs/gst/sdp/Makefile.am:
136 * gst-libs/gst/tag/Makefile.am:
137 * gst-libs/gst/video/Makefile.am:
138 introspection: Build pkgconfig before all libraries and set PKG_CONFIG_PATH
139 This way g-ir-scanner can find the gstreamer-*-0.10 pkg-config files.
141 2009-09-12 02:23:07 +0100 Jan Schmidt <thaytan@noraisin.net>
143 * ext/theora/theoraenc.c:
144 theoraenc: Fix a string leak in _getcaps()
146 2009-09-11 23:49:11 +0100 Jan Schmidt <thaytan@noraisin.net>
183 0.10.24.2 pre-release
185 2009-09-11 21:44:18 +0100 Jan Schmidt <thaytan@noraisin.net>
187 * tests/check/elements/audioresample.c:
188 check: Improve audioresample test
189 Make the audioresample test work with CK_FORK=no, and
190 turn a g_print into a GST_INFO.
192 2009-09-11 22:09:06 +0200 Benjamin Otte <otte@gnome.org>
194 * gst/videotestsrc/videotestsrc.c:
195 videotestsrc: Fix crashes with even widths
196 The fix for green lines introduced by commit
197 35fdfcc6258c66ba462a4330a35deffb0f2b501d caused invalid memory accesses
198 for even widths. This patch fixes it.
200 2009-09-11 15:11:41 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
202 * gst/playback/gstplaybin2.c:
203 playbin2: Implement GstStreamVolume interface
205 2009-09-11 15:04:42 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
207 * gst/volume/gstvolume.c:
208 * gst/volume/gstvolume.h:
209 * tests/check/Makefile.am:
210 * tests/check/elements/volume.c:
211 volume: Implement GstStreamVolume interface
213 2009-09-11 14:54:17 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
215 * docs/libs/gst-plugins-base-libs-docs.sgml:
216 * docs/libs/gst-plugins-base-libs-sections.txt:
217 * gst-libs/gst/interfaces/Makefile.am:
218 * gst-libs/gst/interfaces/streamvolume.c:
219 * gst-libs/gst/interfaces/streamvolume.h:
220 * gst/playback/Makefile.am:
221 * win32/common/libgstinterfaces.def:
222 interfaces: API: Add GstStreamVolume interface
225 2009-09-11 12:20:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
227 * gst-libs/gst/rtsp/gstrtspconnection.c:
228 rtsp: properly fix the HTTP manual mode
229 When we're not parsing HTTP, return EPARSE when we get an HTTP
232 2009-09-11 10:16:15 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
234 * gst-libs/gst/interfaces/mixertrack.h:
235 mixertrack: add READONLY and WRITEONLY flags
236 Should really have been READABLE and WRITABLE, but those are hard to
237 add whilst maintaining backwards compatibility. See #343615.
238 API: GST_MIXER_TRACK_READONLY
239 API: GST_MIXER_TRACK_WRITEONLY
241 2009-09-11 10:02:54 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
243 * gst-libs/gst/audio/gstringbuffer.c:
244 ringbuffer: fix build against core that has debugging disabled
245 The macro is called GST_DISABLE_GST_DEBUG, not GST_DISABLE_DEBUG.
247 2009-09-11 07:38:28 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
249 * gst/videorate/gstvideorate.c:
250 videorate: Add Since marker for the new skip-to-first property
252 2009-09-11 07:36:10 +0200 Olivier Crête <olivier.crete@collabora.co.uk>
254 * gst/videorate/gstvideorate.c:
255 * gst/videorate/gstvideorate.h:
256 videorate: Make videorate work with a live source
257 Add a property that makes videorate skip to the first buffer it
258 receives instead of padding the stream from segment start to the
262 2009-09-11 07:20:49 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
264 * gst-libs/gst/fft/gstfft.h:
265 * gst-libs/gst/fft/gstfftf32.h:
266 * gst-libs/gst/fft/gstfftf64.h:
267 * gst-libs/gst/fft/gstffts16.h:
268 * gst-libs/gst/fft/gstffts32.h:
269 fft: Mark one function as const and add notes that the structs should be private in 0.11
271 2009-09-10 22:28:19 +0300 Stefan Kost <ensonic@users.sf.net>
273 * gst-libs/gst/audio/gstringbuffer.c:
274 ringbuffer: add human readable format names when logging
275 Add string array with human readable names for format and type to be used in log
278 2009-09-10 18:19:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
280 * gst-libs/gst/rtp/gstbasertppayload.c:
281 basertppay: don't print RTP timestamps as clocktime
282 Don't try to print the RTP timestamp as a GstClockTime, it's just a guint32.
285 2009-09-10 16:55:31 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
287 * gst/playback/gstplaybin.c:
288 * gst/playback/gstplaybin2.c:
289 playbin(2): Document that the volume property uses a linear scale
292 2009-09-10 14:04:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
294 * gst-libs/gst/rtsp/gstrtspconnection.c:
295 rtsp: don't return EPARSE
296 Don't blindly return EPARSE when http mode is disabled.
297 Restore old http mode after temporarily setting it to TRUE.
299 2009-09-10 12:38:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
301 * gst-libs/gst/audio/gstbaseaudiosink.c:
302 baseaudiosink: add ugly backward compat hack
303 Check for pulsesink < 0.10.17 because it includes code that is now included in
304 baseaudiosink. Disable that code in baseaudiosink to be compatible with the
307 2009-09-10 10:56:29 +0200 Benjamin Otte <otte@gnome.org>
309 * gst/ffmpegcolorspace/imgconvert.c:
310 ffmpegcolorspace: Handle YVU9/YUV9 conversion with odd widths
311 A green border could be visible when converting to Y444 or RGB, because
312 the last chroma samples weren't copied correctly
314 2009-09-10 10:43:37 +0200 Benjamin Otte <otte@gnome.org>
316 * gst/videotestsrc/videotestsrc.c:
317 videotestsrc: Fix YVU9 and YUV9
318 - Buffer sizes were computed different from ffmpegcolorspace
319 - Green bar on right size for widths not divisable by 4
321 2009-09-10 10:08:28 +0200 Benjamin Otte <otte@gnome.org>
323 * gst/videotestsrc/videotestsrc.c:
324 videotestsrc: Fix image for odd widths in some formats
325 videotestsrc rounds chroma down. This causes it to omit the last chroma
326 value completely for odd widths when the chroma is downsampled.
327 This patch special cases the last pixel to not be rounded down.
329 2009-09-10 10:02:58 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
331 * ext/ogg/gstoggdemux.c:
332 oggdemux: Handle kate and cmml as sparse streams too
334 2009-09-10 10:00:16 +0200 Tim-Philipp Müller <tim.muller@collabora.co.uk>
336 * ext/ogg/gstoggdemux.c:
337 * ext/ogg/gstoggdemux.h:
338 oggdemux: Better handling of sparse streams by sending segment updates
341 2009-09-10 09:43:28 +0300 Stefan Kost <ensonic@users.sf.net>
343 * gst/playback/gsturidecodebin.c:
344 docs: tell a biit more about uri-decodebin and buffering
346 2009-09-09 18:24:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
348 * gst-libs/gst/audio/gstbaseaudiosink.c:
349 baseaudiosink: take clock time in setcaps
350 Take the time of the clock so that the last_time field is set. This is important
351 for sinks that restart their internal ringbuffer after a caps change and need to
352 know the last know position.
354 2009-09-09 18:24:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
356 * gst-libs/gst/audio/gstaudioclock.c:
357 audioclock: add some more debug
359 2009-09-09 16:44:24 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
361 * ext/theora/theoraenc.c:
362 theoraenc: Print a debug message with supported formats
364 2009-09-07 17:29:38 +0200 Benjamin Otte <otte@gnome.org>
366 * ext/theora/theoraenc.c:
367 theora: Check supported input formats in getcaps function
368 We want to fail early when an older libtheora release is used that does
369 not support Y444 or Y42B formats, so use a getcaps function that does
372 2009-09-04 21:37:04 +0200 Benjamin Otte <otte@gnome.org>
374 * ext/theora/theoraenc.c:
375 theora: Implement support in theoraenc for Y444 and Y42B
378 2009-09-04 20:23:52 +0200 Benjamin Otte <otte@gnome.org>
380 * ext/theora/theoraenc.c:
381 theora: Refactor the buffer copy code
383 2009-09-04 16:59:49 +0200 Benjamin Otte <otte@gnome.org>
385 * ext/theora/theoraenc.c:
386 theora: Split yuv_buffer creation into its own function
388 2009-09-04 16:49:08 +0200 Benjamin Otte <otte@gnome.org>
390 * ext/theora/theoraenc.c:
391 theora: Split out buffer resize in its own function
393 2009-09-04 14:06:09 +0200 Benjamin Otte <otte@gnome.org>
395 * ext/theora/theoraenc.c:
396 theora: Add assertions that functions don't fail
397 Some functions in libtheora can return an error, but that error cannot
398 ever happen inside theoraenc. In those cases assert that it doesn't.
400 2009-09-09 16:21:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
402 * tests/examples/seek/seek.c:
403 seek: make stop state configurable
404 Make it easy to experiment with different stop states (NULL and READY)
406 2009-09-09 16:19:32 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
408 * gst-libs/gst/audio/gstbaseaudiosink.c:
409 baseaudiosink: correct for clock reset
410 When going to NULL, we reset the ringbuffer so that it starts beck from 0. We
411 also make sure that the clock is updated with the elapsed time so that it
412 alsways increments even when the ringbuffer goes back to 0. When this happened
413 we need to adjust the sample position for the reset ringbuffer.
416 2009-09-09 16:17:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
418 * gst-libs/gst/audio/gstbaseaudiosink.h:
419 baseaudiosink: whitespace fixes
421 2009-09-09 16:16:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
423 * gst-libs/gst/audio/gstringbuffer.c:
424 ringbuffer: add more debug
426 2009-09-09 10:25:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
428 * gst-libs/gst/interfaces/colorbalance.h:
429 * gst-libs/gst/interfaces/mixer.h:
432 2009-09-08 17:59:30 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
434 * gst-libs/gst/video/gstvideosink.c:
435 * gst-libs/gst/video/gstvideosink.h:
436 videosink: add "show-preroll-frame" property
437 Add a property to disable rendering of video frames during preroll. This
438 will only work for videosinks that use the new ::show_frame() vfunc instead
439 of overriding basesink's preroll and render vfuncs directly.
440 API: GstVideoSink:show-preroll-frame
442 2009-09-08 17:43:26 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
444 * sys/ximage/ximagesink.c:
445 * sys/xvimage/xvimagesink.c:
446 ximagesink, xvimagesink: use new GstVideoSink::show_frame() vfunc
448 2009-09-08 18:19:19 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
450 * gst-libs/gst/video/gstvideosink.c:
451 * gst-libs/gst/video/gstvideosink.h:
452 video: add GstVideoSinkClass::show_frame()
453 Add ::show_frame() vfunc which maps to basesink's ::preroll and ::render
454 vfuncs and add some gtk-doc chunks.
455 API: GstVideoSinkClass::show_frame()
457 2009-09-08 16:00:47 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
459 * gst-libs/gst/interfaces/navigation.c:
460 navigation: don't do stuff inside g_return_val_if_fail() statements
461 Or it will all fall apart if someone compiles with -DG_DISABLE_ASSERT.
463 2009-08-31 20:24:22 +0200 Havard Graff <havard.graff@tandberg.com>
465 * gst-libs/gst/interfaces/navigation.c:
466 navigation: Fix compiler warning with MSVC
469 2009-08-31 20:31:56 +0200 Havard Graff <havard.graff@tandberg.com>
471 * gst-libs/gst/rtp/gstbasertpdepayload.c:
472 basertpdepayload: fix event forwarding
474 2009-08-31 20:36:37 +0200 Havard Graff <havard.graff@tandberg.com>
476 * gst-libs/gst/rtp/gstrtcpbuffer.c:
477 rtcpbuffer: add missing break in handling of GST_RTCP_TYPE_PSFB
480 2009-09-08 13:02:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
482 * gst/playback/gstplaybin2.c:
483 * gst/playback/gstplaysink.c:
484 * gst/playback/gstplaysink.h:
487 2009-09-08 12:59:20 +0200 Håvard Graff <havard.graff@tandberg.com>
489 * gst-libs/gst/audio/gstbaseaudiosrc.c:
490 baseaudiosrc: improve slave skew resync
491 The old one did the mistake of not actually advancing the ringbuffer, it just
492 adjusted the segbase, introducing the whole lenght of the ringbuffer as an
493 extra delay in the pipeline.
494 Also make sure that the resync can never go back in time, producing the same
495 timestamps that has already been produced, as this can cause severe problems
496 for sinks and other synching mechanisms.
499 2009-09-07 17:13:12 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
501 * gst/typefind/gsttypefindfunctions.c:
502 typefinding: disable typefinder for headerless flac
503 Disable headerless flac typefinder as long as it happily typefinds anything
504 including /dev/urandom as flac and as long as it's not particularly useful
505 given that such streams don't really exist in the wild.
506 Also fix up some comments so that gtk-doc doesn't complain about them.
508 2009-09-06 15:21:43 +0300 René Stadler <mail@renestadler.de>
510 * sys/ximage/ximagesink.c:
511 ximagesink: fix small memory leak when setting window title
513 2009-09-06 01:42:42 +0300 René Stadler <mail@renestadler.de>
515 * sys/xvimage/xvimagesink.c:
516 xvimagesink: fix small memory leak when setting window title
518 2009-09-05 13:55:27 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
521 introspection: Add *.gir and *.typelib to .gitignore
523 2009-09-05 13:46:58 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
525 * gst-libs/gst/app/Makefile.am:
526 * gst-libs/gst/audio/Makefile.am:
527 * gst-libs/gst/interfaces/Makefile.am:
528 * gst-libs/gst/pbutils/Makefile.am:
529 * gst-libs/gst/rtsp/Makefile.am:
530 * gst-libs/gst/video/Makefile.am:
531 introduction: Fix out-of-tree build
533 2009-09-05 13:13:23 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
535 * gst-libs/gst/rtsp/Makefile.am:
536 rtsp: Fix introspection build by ordering sources/headers in dependency order
538 2009-09-05 13:09:17 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
540 * gst-libs/gst/audio/Makefile.am:
541 audio: Remove debug echo
543 2009-09-05 13:08:19 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
545 * gst-libs/gst/audio/Makefile.am:
546 audio: Fix build of introspection data by using dependency order for the headers/sources
548 2009-09-05 12:31:47 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
550 * gst-libs/gst/app/Makefile.am:
551 * gst-libs/gst/audio/Makefile.am:
552 * gst-libs/gst/cdda/Makefile.am:
553 * gst-libs/gst/fft/Makefile.am:
554 * gst-libs/gst/interfaces/Makefile.am:
555 * gst-libs/gst/netbuffer/Makefile.am:
556 * gst-libs/gst/pbutils/Makefile.am:
557 * gst-libs/gst/riff/Makefile.am:
558 * gst-libs/gst/rtp/Makefile.am:
559 * gst-libs/gst/rtsp/Makefile.am:
560 * gst-libs/gst/sdp/Makefile.am:
561 * gst-libs/gst/tag/Makefile.am:
562 * gst-libs/gst/video/Makefile.am:
563 introspection: Strip Gst prefix from all types/functions
565 2009-09-05 11:49:41 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
567 * gst-libs/gst/Makefile.am:
568 * gst-libs/gst/app/Makefile.am:
569 * gst-libs/gst/audio/Makefile.am:
570 * gst-libs/gst/fft/Makefile.am:
571 * gst-libs/gst/interfaces/Makefile.am:
572 * gst-libs/gst/netbuffer/Makefile.am:
573 * gst-libs/gst/pbutils/Makefile.am:
574 * gst-libs/gst/riff/Makefile.am:
575 * gst-libs/gst/rtp/Makefile.am:
576 * gst-libs/gst/rtsp/Makefile.am:
577 * gst-libs/gst/sdp/Makefile.am:
578 * gst-libs/gst/tag/Makefile.am:
579 * gst-libs/gst/video/Makefile.am:
580 introspection: Fix build if gir-repository is not installed
582 2009-09-05 11:37:14 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
584 * gst-libs/gst/video/Makefile.am:
585 video: Add gobject-introspection support
587 2009-09-05 11:35:34 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
589 * gst-libs/gst/tag/Makefile.am:
590 tag: Add gobject-introspection support
592 2009-09-05 11:34:11 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
594 * gst-libs/gst/sdp/Makefile.am:
595 sdp: Add gobject-introspection support
597 2009-09-05 11:31:48 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
599 * gst-libs/gst/app/Makefile.am:
600 * gst-libs/gst/audio/Makefile.am:
601 * gst-libs/gst/interfaces/Makefile.am:
602 * gst-libs/gst/pbutils/Makefile.am:
603 libs: Add nodist headers and sources to the introspection files
605 2009-09-05 11:28:59 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
607 * gst-libs/gst/rtsp/Makefile.am:
608 rtsp: Add gobject-introspection support
610 2009-09-05 11:25:42 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
612 * gst-libs/gst/rtp/Makefile.am:
613 rtp: Add gobject-introspection support
615 2009-09-05 11:23:13 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
617 * gst-libs/gst/riff/Makefile.am:
618 riff: Add gobject-introspection support
620 2009-09-05 11:20:51 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
622 * gst-libs/gst/pbutils/Makefile.am:
623 pbutils: Add gobject-introspection support
625 2009-09-05 11:17:07 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
627 * gst-libs/gst/netbuffer/Makefile.am:
628 netbuffer: Add gobject-introspection support
630 2009-09-05 11:15:05 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
632 * gst-libs/gst/interfaces/Makefile.am:
633 interfaces: Add gobject-introspection support
635 2009-09-05 11:04:19 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
637 * gst-libs/gst/fft/Makefile.am:
638 fft: Add gobject-introspection support
640 2009-09-05 11:01:44 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
642 * gst-libs/gst/cdda/Makefile.am:
643 cdda: Add gobject-introspection support
644 This is disabled for now until gobject-introspection is fixed
646 2009-09-05 10:50:48 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
648 * gst-libs/gst/audio/Makefile.am:
649 audio: Add gobject-introspection support
651 2009-09-05 10:40:21 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
654 * gst-libs/gst/app/Makefile.am:
655 app: Add gobject-introspection support
657 2009-09-05 10:20:24 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
660 Automatic update of common submodule
661 From 00a859e to 19fa4f3
663 2009-09-04 15:48:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
665 * gst/typefind/gsttypefindfunctions.c:
666 typefind: fix midi typefinding
667 We already have a audio/midi typefinder so don't override it with the midi in
668 RIFF typefinder or else we fail to detect plain midi files.
670 2009-09-04 11:29:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
672 * gst/playback/gsturidecodebin.c:
673 uridecodebin: do buffering for more uris
674 Add ssh://, ftp://, sftp://, myth:// to the list of uris that require
678 2009-09-04 07:36:10 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
680 * gst/typefind/gsttypefindfunctions.c:
681 typefindfunctions: Add typefinder for Midi inside RIFF
682 This is a standard Midi file format that should be supported by
683 all Midi decoders and also has the mimetype audio/mid according to
684 the Midi specification homepage.
687 2009-09-03 18:53:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
689 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
690 audiortppay: add some debugging
692 2009-09-03 17:53:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
694 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
695 audiortppay: handle gaps
696 Add various conversion functions between time<->bytes<->rtptime that will be
698 Refactor the min/max packet length code so that it can be used for both
699 sample/frame based payloaders. Cache the returned values.
701 When we discover a DISCONT buffer, make the outgoing RTP timestamps have the
702 same gap as the GStreamer timestamps gap.
704 2009-09-03 14:13:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
706 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
707 audiortppay: fix frame duration calculations
708 Fix the calculation of the frame duration and rtp timestamps.
711 2009-09-03 14:13:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
713 * gst-libs/gst/rtp/gstbasertppayload.c:
714 rtppay: add some debugging
716 2009-09-02 19:49:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
718 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
719 audiortppay: use offsets for RTP timestamps
720 Have a custom sample/frame function to generate an offset that the base class
721 will use for generating RTP timestamps. This results in perfect RTP timestamps
722 on the output buffers.
723 Refactor setting metadata on output buffers.
724 Add some more functionality to _flush().
725 Handle DISCONT on the input buffers and set the marker bit and DISCONT flag on
726 the next outgoing buffer.
727 Flush the pending data on EOS.
729 2009-09-02 13:13:54 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
731 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
732 audiortppay: move function around
734 2009-09-02 13:12:28 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
736 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
737 audiortppay: fix sample duration calculation
739 2009-09-02 12:24:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
741 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
742 audiortppay: more refactoring
743 Unify the sample/frame buffer handling code by making the functions plugable.
745 2009-09-02 12:03:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
747 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
748 * gst-libs/gst/rtp/gstbasertpaudiopayload.h:
749 audiortppayload: refactor some more
750 Refactor getting the packet min/max size and alignment code.
751 Refactor converting bytes to time.
752 change some variable to something shorter.
754 2009-09-02 10:46:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
756 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
757 * gst-libs/gst/rtp/gstbasertpaudiopayload.h:
758 * win32/common/libgstrtp.def:
759 audiortppayload: refactor and cleanup
760 Always use the adapter when we need to fragment the incomming buffer. Use more
761 modern adapter functions to avoid malloc and memcpy. The overall result is that
762 the code looks cleaner while it should be equally fast and in some case avoid a
764 Use the adapter timestamping functions for more precise timestamps in case of
766 Cache some values instead of recalculating them.
767 Add gst_base_rtp_audio_payload_flush() to flush a certain amount of bytes from
768 the internal adapter.
769 API: GstBaseRTPAudioPayload::gst_base_rtp_audio_payload_flush()
771 2009-09-03 16:56:55 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
776 2009-09-03 11:29:23 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
778 * gst-libs/gst/rtp/gstbasertppayload.c:
779 basertppay: add property to disable perfect RTP time
780 Add a property to disable the generation of perfect RTP timestamps. By default
782 API: GstBaseRTPPayload::perfect-rtptime
784 2009-09-02 19:47:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
786 * gst-libs/gst/rtp/gstbasertppayload.c:
787 basertppay: allow subclasses to influence RTP time
788 Allow subclasses to use the OFFSET field on RTP buffers to influence the way in
789 which RTP timestamps are generated. Usually timestamps are created from the
790 GStreamer timestamps on the buffer, which could result in imperfect RTP
793 2009-09-02 19:44:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
795 * gst-libs/gst/rtp/gstbasertppayload.h:
796 basertppay: add macro to cast
798 2009-09-01 18:26:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
800 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
801 audiopayload: code cleanups
803 2009-09-01 18:08:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
805 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
806 audiortppayload: don't check adapter
807 the adapter is never NULL so we don't need to check it.
808 Use _scale functions to avoid overflows.
810 2009-09-03 00:14:02 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
813 * gst/typefind/Makefile.am:
814 * gst/typefind/gsttypefindfunctions.c:
815 typefinding: move gio-based xdg mime typefinder from -bad to -base
816 Its purposes is mainly to avoid false positives (e.g. mp3 typefinder
817 reporting a 20% probability and somesuch). Won't be registered if
818 the gio plugin has been disabled via ./configure --disable-gio.
820 2009-09-01 15:06:51 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
822 * gst/subparse/gstsubparse.c:
823 subparse: GstAdapter is not a GstObject and should be freed with g_object_unref
825 2009-09-01 15:02:37 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
827 * sys/v4l/v4lsrc_calls.c:
828 v4lsrc: fix timestamping for when we do not have a clock yet
831 2009-09-01 14:30:41 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
833 * sys/v4l/v4lsrc_calls.c:
834 v4lsrc: don't log not-yet-initialised integer value
836 2009-09-01 14:28:48 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
838 * sys/v4l/v4lsrc_calls.c:
839 v4lsrc: avoid unnecessary run-time type checks in custom buffer finalize
840 And reflow code to be more indent friendly.
842 2009-09-01 10:39:52 +0200 Jonas Holmberg <jonas.holmberg@axis.com>
844 * gst-libs/gst/rtp/gstbasertppayload.c:
845 * gst-libs/gst/rtp/gstbasertppayload.h:
846 basertppayload: Make instance init faster by not reading /dev/urandom 3 times
847 ... which is the default seed when creating a new GRand. Because
848 GLib in older versions used buffered IO this would take a lot of time.
849 Instead use the global GRand for getting random numbers and keep the
850 three instance GRand for backward compatibility with a simple seed.
853 2009-08-31 22:48:01 +0300 Stefan Kost <ensonic@users.sf.net>
855 * gst/adder/gstadder.c:
856 adder: improve caps filter functionality. Fixes #590146.
857 Also use the capsfilter if there is no src-peer as the caps constrain what
858 we can do. Don't create any_caps as a default, as we check for NULL to skip the
859 filtering. This is a (small) performance regression as we always intersect
862 2009-08-31 11:10:55 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
864 * gst/playback/gstdecodebin2.c:
865 decodebin2: Post missing plugin messages before any error messages
867 2009-08-28 19:06:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
869 * gst-libs/gst/cdda/gstcddabasesrc.c:
870 cddabasesrc: safely handle the indexes
872 2009-08-28 19:06:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
874 * win32/common/libgstrtsp.def:
875 def: add new rtsp symbols
877 2009-08-28 14:08:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
879 * gst-libs/gst/rtp/gstbasertppayload.h:
880 basertppayload: whitespace fixes.
882 2009-08-27 18:59:49 +0200 Marc-André Lureau <mlureau@flumotion.com>
884 * gst/gdp/gstgdppay.c:
885 Bug 593035 - set IN_CAPS for streamheader buffer
887 2009-08-26 16:56:19 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
889 * gst/playback/gstinputselector.c:
890 * gst/playback/gststreamselector.c:
891 playbin: The internally linked pad of the selector might be NULL in some cases
893 2009-08-26 16:45:49 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
895 * gst/playback/gstinputselector.c:
896 * gst/playback/gststreamselector.c:
897 playbin: Fix iterate internal linked pads functions for the stream selectors
898 This now used the new gst_iterator_new_single() function and as a side effect
901 2009-08-26 09:08:53 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
903 * gst-libs/gst/riff/riff-ids.h:
904 * gst-libs/gst/riff/riff-read.c:
905 riff: Add support for AVF files
906 AVF is valid RIFF but has AVF0 has first fourcc instead of RIFF.
909 2009-08-26 09:08:12 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
911 * gst/typefind/gsttypefindfunctions.c:
912 typefindfunctions: Detect AVF files as RIFF files too
913 AVF is valid RIFF but has AVF0 as first fourcc instead of RIFF.
914 Partially fixes bug #593117.
916 2009-08-21 11:51:47 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
918 * tests/check/elements/audioresample.c:
919 audioresample: Add unit test for checking for timestamp drifts
920 This also checks for perfect timestamping and offsetting.
922 2009-08-21 10:11:23 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
924 * gst/audioresample/gstaudioresample.c:
925 audioresample: Fix drain processing
926 In case we have to convert internally don't process output length input samples
927 but history length input samples.
929 2009-08-21 10:02:05 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
931 * tests/check/elements/audioresample.c:
932 audioresample: Improve debugging a bit in the unit test
934 2009-08-21 10:00:49 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
936 * gst/audioresample/gstaudioresample.c:
937 audioresample: On the first buffer we need discont handling
938 Otherwise we won't get upstream timestamps and everything and all
939 output buffers would have -1 timestamps.
941 2009-08-21 08:23:39 +0400 Руслан Ижбулатов <lrn1986@gmail.com>
944 * gst/subparse/gstsubparse.c:
945 subparse: Remove dependency on regex.h as it's not used anyway
948 2009-08-21 06:58:31 +0200 Kipp Cannon <kcannon@ligo.caltech.edu>
950 * gst/audioresample/gstaudioresample.c:
951 audioresample: Fix buffer overflow when pushing the drain
953 2009-08-21 06:57:58 +0200 Kipp Cannon <kcannon@ligo.caltech.edu>
955 * gst/audioresample/gstaudioresample.c:
956 * gst/audioresample/gstaudioresample.h:
957 audioresample: Fix timestamp drift
960 2009-08-24 11:34:35 -0700 David Schleef <ds@schleef.org>
962 * ext/gnomevfs/gstgnomevfssrc.c:
963 * ext/ogg/gstogmparse.c:
964 * ext/pango/gsttextrender.c:
965 * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
966 * gst/playback/gstinputselector.c:
967 * gst/playback/gststreamselector.c:
968 * gst/subparse/gstsubparse.c:
969 * sys/v4l/gstv4lmjpegsink.c:
970 * sys/v4l/gstv4lmjpegsrc.c:
971 * sys/v4l/gstv4lsrc.c:
972 Remove Ronald Bultje from Authors field
973 Replaced with "GStreamer maintainers
974 <gstreamer-devel@lists.sourceforge.net>" or just removed,
975 depending on the number of other authors.
977 2009-08-24 15:06:28 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
979 * gst/playback/gstplaybin2.c:
980 playbin2: fix refcounting of _get_sink()
981 g_value_set_object() increases the refcount of the sink, which is not needed
982 because the object should already be refcounted. Make sure this is always the
983 case and use g_value_take_object().
986 2009-08-24 14:39:16 +0200 Peter Kjellerstedt <pkj@axis.com>
988 * gst-libs/gst/rtsp/gstrtspdefs.c:
989 rtsp: Mark Transport as supporting multiple values.
991 2009-08-24 13:58:17 +0200 Peter Kjellerstedt <pkj@axis.com>
993 * gst-libs/gst/rtsp/gstrtspconnection.h:
994 * gst-libs/gst/rtsp/gstrtspdefs.h:
995 * gst-libs/gst/rtsp/gstrtspmessage.h:
996 rtsp: Added missing Since tags.
998 2009-08-24 13:27:55 +0200 Eero Nurkkala <ext-eero.nurkkala at nokia.com>
1000 * gst-libs/gst/audio/gstringbuffer.c:
1001 ringbuffer: Improve audiosink startup performance
1002 When we start the ringbuffer, immediatly continue processing samples if the
1003 writer prepared some for us.
1006 2009-08-17 11:53:43 +0200 Peter Kjellerstedt <pkj@axis.com>
1008 * gst-libs/gst/rtsp/gstrtspconnection.c:
1009 * gst-libs/gst/rtsp/gstrtspconnection.h:
1010 rtsp: Added new API for sending using GstRTSPWatch.
1011 The new API to send messages using GstRTSPWatch will first try to send the
1012 message immediately. Then, if that failed (or the message was not sent
1013 fully), it will queue the remaining message for later delivery. This avoids
1014 unnecessary context switches, and makes it possible to keep track of
1015 whether the connection is blocked (the unblocking of the connection is
1016 indicated by the reception of the message_sent signal).
1017 This also deprecates the old API (gst_rtsp_watch_queue_data() and
1018 gst_rtsp_watch_queue_message().)
1019 API: gst_rtsp_watch_write_data()
1020 API: gst_rtsp_watch_send_message()
1022 2009-08-17 11:46:32 +0200 Peter Kjellerstedt <pkj@axis.com>
1024 * gst-libs/gst/rtsp/gstrtspconnection.c:
1025 rtsp: Made gst_rtsp_watch_queue_data() thread safe.
1027 2009-06-17 15:37:53 +0200 Peter Kjellerstedt <pkj@axis.com>
1029 * gst-libs/gst/rtsp/gstrtspconnection.c:
1030 * gst-libs/gst/rtsp/gstrtspconnection.h:
1031 rtsp: Added gst_rtsp_connection_set_http_mode().
1032 With gst_rtsp_connection_set_http_mode() it is possible to tell the
1033 connection whether to allow HTTP messages to be supported. By enabling HTTP
1034 support the automatic HTTP tunnel support will also be disabled.
1035 API: gst_rtsp_connection_set_http_mode()
1037 2009-06-16 19:35:23 +0200 Peter Kjellerstedt <pkj@axis.com>
1039 * gst-libs/gst/rtsp/gstrtspconnection.c:
1040 rtsp: Allow gst_rtsp_connection_do_tunnel() to just setup decoding context.
1041 If the second connection passed to gst_rtsp_connection_do_tunnel() is NULL
1042 then just setup the base64 decoding context for the first connection.
1044 2009-06-16 19:04:54 +0200 Peter Kjellerstedt <pkj@axis.com>
1046 * gst-libs/gst/rtsp/gstrtspconnection.c:
1047 rtsp: Write as much as possible in gst_rtsp_source_dispatch().
1048 Try to write as much as possible if there are multiple messages queued.
1050 2009-06-16 18:38:02 +0200 Peter Kjellerstedt <pkj@axis.com>
1052 * gst-libs/gst/rtsp/gstrtspconnection.c:
1053 * gst-libs/gst/rtsp/gstrtspconnection.h:
1054 rtsp: Add error_full callback to GstRTSPWatchFuncs.
1055 The error_full callback is similar to the error callback, but allows for
1056 better error handling. For read errors a partial message is provided to
1057 help an RTSP server generate a more correct error response, and for write
1058 errors the write queue id of the failed message is returned.
1060 2009-08-17 18:29:17 +0200 Peter Kjellerstedt <pkj@axis.com>
1062 * gst-libs/gst/rtsp/gstrtspconnection.c:
1063 rtsp: Made read_line() support LWS.
1064 Rewrote read_line() to support LWS (Line White Space), the method used by
1065 RTSP (and HTTP) to break long lines. Also added support for \r and \n as
1066 line endings (in addition to the official \r\n).
1068 2009-08-20 14:12:50 +0200 Peter Kjellerstedt <pkj@axis.com>
1070 * gst-libs/gst/rtsp/gstrtspconnection.c:
1071 * gst-libs/gst/rtsp/gstrtspdefs.c:
1072 * gst-libs/gst/rtsp/gstrtspdefs.h:
1073 rtsp: Do not split headers which should not be split.
1074 From RFC 2068 section 4.2: "Multiple message-header fields with the same
1075 field-name may be present in a message if and only if the entire
1076 field-value for that header field is defined as a comma-separated list
1077 [i.e., #(values)]." This means that we should not split other headers which
1078 may contain a comma, e.g., Range and Date.
1080 2009-08-20 14:12:09 +0200 Peter Kjellerstedt <pkj@axis.com>
1082 * gst-libs/gst/rtsp/gstrtspconnection.c:
1083 rtsp: Parse WWW-Authenticate headers correctly.
1084 Due to the odd syntax for WWW-Authenticate (and Proxy-Authenticate) which
1085 allows commas both to separate between multiple challenges, and within the
1086 challenges themself, we need to take some extra care to split these headers
1089 2009-06-17 21:46:27 +0200 Peter Kjellerstedt <pkj@axis.com>
1091 * gst-libs/gst/rtsp/gstrtspconnection.c:
1092 rtsp: Improve parse_line().
1093 Make parse_line() handle keys with multiple values on one line correctly.
1095 2009-06-17 23:15:23 +0200 Peter Kjellerstedt <pkj@axis.com>
1097 * gst-libs/gst/rtsp/gstrtspconnection.c:
1098 rtsp: Rewrote setup_tunneling().
1099 Rewrote setup_tunneling() to use normal GstRTSPMessages instead of hard
1100 coded strings and duplicates of the message parsing code.
1102 2009-08-24 10:20:16 +0200 Peter Kjellerstedt <pkj@axis.com>
1104 * gst-libs/gst/rtsp/gstrtspconnection.c:
1105 * gst-libs/gst/rtsp/gstrtspdefs.c:
1106 * gst-libs/gst/rtsp/gstrtspdefs.h:
1107 rtsp: Rewrote gen_tunnel_reply().
1108 Rewrote gen_tunnel_reply() to generate a normal GstRTSPMessage rather
1109 than a hard coded string.
1111 2009-08-24 10:19:35 +0200 Peter Kjellerstedt <pkj@axis.com>
1113 * gst-libs/gst/rtsp/gstrtspconnection.c:
1114 rtsp: Ignore the Content-Length for POST requests.
1115 The Content-Length for POST requests with an x-sessioncookie header should
1116 be ignored as the length is bogus and only there to fool proxies.
1118 2009-06-17 20:52:48 +0200 Peter Kjellerstedt <pkj@axis.com>
1120 * gst-libs/gst/rtsp/gstrtspconnection.c:
1121 rtsp: Normalize lines (remove extra whitespace) before parsing.
1123 2009-06-10 13:11:31 +0200 Peter Kjellerstedt <pkj@axis.com>
1125 * gst-libs/gst/rtsp/gstrtspconnection.c:
1126 rtsp: Made parse_string() return a result.
1127 This will catch parsing errors when a too long string is received.
1129 2009-06-10 11:43:31 +0200 Peter Kjellerstedt <pkj@axis.com>
1131 * gst-libs/gst/rtsp/gstrtspconnection.c:
1132 rtsp: Improved parsing of messages.
1133 Do not abort message parsing as soon as there is an error. Instead parse
1134 as much as possible to allow a server to return as meaningful an error as
1137 2009-06-09 17:54:20 +0200 Peter Kjellerstedt <pkj@axis.com>
1139 * gst-libs/gst/rtsp/gstrtspconnection.c:
1140 * gst-libs/gst/rtsp/gstrtspdefs.c:
1141 * gst-libs/gst/rtsp/gstrtspdefs.h:
1142 * gst-libs/gst/rtsp/gstrtspmessage.c:
1143 * gst-libs/gst/rtsp/gstrtspmessage.h:
1144 rtsp: Added support for HTTP messages
1146 2009-06-09 16:22:17 +0200 Peter Kjellerstedt <pkj@axis.com>
1148 * gst-libs/gst/rtsp/gstrtspconnection.c:
1149 * gst-libs/gst/rtsp/gstrtspconnection.h:
1150 rtsp: Added gst_rtsp_connection_create_from_fd().
1151 API: gst_rtsp_connection_create_from_fd()
1153 2009-06-09 15:27:17 +0200 Peter Kjellerstedt <pkj@axis.com>
1155 * gst-libs/gst/rtsp/gstrtspconnection.c:
1156 rtsp: Add initial buffer support.
1157 The initial buffer contains data for a connection which should be used
1158 before starting to actually read anything from the socket.
1160 2009-08-24 13:15:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1162 * gst-libs/gst/app/gstappsink.c:
1163 appsink: don't block in paused
1164 When we are asked to unlock we should either leave the render function or call
1165 the wait_preroll method to release the stream lock.
1168 2009-08-24 13:06:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1170 * docs/libs/gst-plugins-base-libs-sections.txt:
1171 docs: fix includes for appsrc/appsink
1173 2009-08-24 11:24:27 +0200 Peter Kjellerstedt <pkj@axis.com>
1175 * gst-libs/gst/rtsp/gstrtspdefs.c:
1176 * gst-libs/gst/rtsp/gstrtspdefs.h:
1177 rtsp: Add support for the Authentication-Info header.
1178 The Authentication-Info header is defined in RFC 2617 (Digest Access
1181 2009-08-20 13:11:07 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
1183 * ext/ogg/gstoggmux.c:
1184 * tests/check/pipelines/oggmux.c:
1185 oggmux: don't drop the streamheader field from the output caps
1186 Revert previous 'fix' for bug #588717 and fix it properly, whilst
1187 maintaining the streamheader field on the output caps. Also make
1188 sure we don't leak header buffers we couldn't push when downstream
1189 is unlinked. Add unit test for the presence of the streamheader
1190 field on the output caps and for the issue from bug #588717.
1192 2009-08-18 21:45:31 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
1194 * gst/playback/gstinputselector.c:
1195 * gst/playback/gststreamselector.c:
1196 streamselector/inputselector: Use iterate internal links instead of deprecated get internal links
1198 2009-08-19 09:31:51 +0200 Peter Kjellerstedt <pkj@axis.com>
1200 * gst-libs/gst/rtsp/gstrtspconnection.c:
1201 rtsp: Avoid duplicated headers.
1202 Remove any existing Session and Date headers before adding new ones
1203 when sending a request. This may happen if the user of this code reuses
1204 a request (rtspsrc does this when resending after authorization fails).
1206 2009-08-18 16:49:58 +0200 Peter Kjellerstedt <pkj@axis.com>
1208 * gst-libs/gst/rtsp/gstrtspconnection.c:
1209 rtsp: Corrected the HTTP digest authorization computation.
1210 Do not use sizeof() on an array passed as an argument to a function and
1211 expect to get anything but the size of a pointer. As a result only the
1212 first 4 (or 8) bytes of the response buffer were initialized to 0 in
1213 auth_digest_compute_response() which caused it to return a string which
1214 was not NUL-terminated...
1216 2009-08-18 11:15:41 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
1218 * gst/playback/gstplaysink.c:
1219 playsink: Also send SEEK events directly to a subpicture sink
1221 2009-08-18 08:39:02 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
1223 * gst/playback/gstplaysink.c:
1224 playsink: If a custom text sink is used, send events to it too
1225 Before, SEEK events would be sent to the video sink, which wouldn't
1226 be linked in any way to the subtitle part of the pipeline and
1227 subparse would never see the SEEK event. This would then seek
1228 the audio/video but the subtitles would continue from the old
1232 2009-08-18 08:20:28 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
1234 * gst/playback/gsturidecodebin.c:
1235 uridecodebin: Make missing plugins emit a warning message, not an error message
1236 The problem with an error message is, that it will stop playback completely
1237 while it could be that only a audio decoder plugin is missing and the video
1238 could be played with the available plugins.
1241 2009-08-13 17:42:07 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
1243 * gst/playback/gsturidecodebin.c:
1244 uridecodebin: Post a correct error message for unknown types
1245 Before we had STREAM/WRONG_TYPE but it's really CORE/MISSING_PLUGIN
1246 because a plugin is missing and nothing else is wrong.
1247 Also make it an error instead of a warning.
1248 Really fixes bug #591677.
1250 2009-08-13 15:48:00 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
1252 * gst/playback/gsturidecodebin.c:
1253 uridecodebin: Post a missing plugin message additional to the error message on unknown types
1256 2009-08-13 10:59:35 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
1258 * gst/playback/gstplaysink.c:
1290 playbin2: fix error message string
1293 2009-08-05 15:38:32 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
1295 * gst-libs/gst/riff/riff-read.c:
1296 riff: align API doc of gst_riff_parse_chunk with reality
1298 2009-08-05 15:36:30 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
1300 * gst/playback/gstdecodebin2.c:
1301 decodebin2: avoid assertion failure on empty/NULL caps
1303 2009-08-12 12:09:45 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
1305 * gst/typefind/gsttypefindfunctions.c:
1306 typefindfunctions: Also detect SVG by the <svg> starting tag
1307 Not all SVG images have the DOCTYPE specified.
1309 2009-08-10 20:18:24 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
1311 * gst-libs/gst/rtsp/gstrtspconnection.c:
1312 rtspconnection: don't use GLib-2.18 function
1313 g_checksum_reset() was added only in GLib 2.18, but we still require
1314 only 2.16, so work around that if we only have 2.16. Fixes #591357.
1316 2009-08-10 15:40:33 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
1318 * tests/check/pipelines/streamheader.c:
1319 streamheader: Fix caps leak in the vorbisenc unit test
1321 2009-08-10 14:14:30 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
1323 * tests/check/pipelines/streamheader.c:
1324 checks: fix stream header unit test hanging in gst_task_cleanup_all()
1325 Set pipelines to NULL state and unref when done.
1327 2009-08-10 10:17:07 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
1329 * gst-libs/gst/rtsp/Makefile.am:
1330 * gst-libs/gst/rtsp/gstrtspconnection.c:
1331 * gst-libs/gst/rtsp/md5.c:
1332 * gst-libs/gst/rtsp/md5.h:
1333 rtsp: Use GLib's GChecksum instead of our own MD5 implementation
1335 2009-08-10 03:46:39 +0300 Mart Raudsepp <leio@gentoo.org>
1337 * gst-libs/gst/interfaces/navigation.c:
1338 navigation: Fix doc blurb typo for gst_navigation_send_key_event
1340 2009-08-09 12:13:16 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
1342 * gst/subparse/gstsubparse.c:
1343 subparse: Allow . instead of , as millisecond delimiter in srt subtitles
1346 2009-08-08 17:51:10 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
1348 * gst-libs/gst/audio/gstaudiosrc.c:
1349 * gst/playback/gstinputselector.c:
1350 * gst/playback/gststreamselector.c:
1351 Revert inlines that cause compiler warnings and are not needed anyway
1353 2009-08-08 15:54:57 +0200 Edward Hervey <bilboed@bilboed.com>
1355 * gst-libs/gst/audio/gstaudioclock.c:
1356 * gst-libs/gst/audio/gstaudiosink.c:
1357 * gst-libs/gst/audio/gstaudiosrc.c:
1358 * gst-libs/gst/audio/gstbaseaudiosrc.c:
1359 * gst-libs/gst/audio/gstringbuffer.c:
1360 * gst-libs/gst/interfaces/propertyprobe.c:
1361 * gst-libs/gst/riff/riff-media.c:
1362 * gst-libs/gst/rtp/gstbasertpdepayload.c:
1363 * gst-libs/gst/video/gstvideofilter.c:
1364 * gst-libs/gst/video/gstvideosink.c:
1365 gst-libs: Remove dead assignments and resulting unused variables.
1367 2009-08-08 15:54:41 +0200 Edward Hervey <bilboed@bilboed.com>
1369 * ext/alsa/gstalsadeviceprobe.c:
1370 * ext/alsa/gstalsasink.c:
1371 * ext/alsa/gstalsasrc.c:
1372 * ext/gnomevfs/gstgnomevfssrc.c:
1373 * ext/ogg/gstoggaviparse.c:
1374 * ext/ogg/gstoggdemux.c:
1375 * ext/ogg/gstoggmux.c:
1376 * ext/pango/gsttextrender.c:
1377 * ext/vorbis/vorbisenc.c:
1378 ext: Remove dead assignments and resulting unused variables.
1380 2009-08-08 15:54:02 +0200 Edward Hervey <bilboed@bilboed.com>
1382 * gst/adder/gstadder.c:
1383 * gst/audioconvert/gstaudioconvert.c:
1384 * gst/audioresample/gstaudioresample.c:
1385 * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
1386 * gst/ffmpegcolorspace/imgconvert.c:
1387 * gst/playback/gstdecodebin.c:
1388 * gst/playback/gstdecodebin2.c:
1389 * gst/playback/gstfactorylists.c:
1390 * gst/playback/gstinputselector.c:
1391 * gst/playback/gstplaysink.c:
1392 * gst/playback/gststreamselector.c:
1393 * gst/tcp/gsttcpclientsink.c:
1394 * gst/videoscale/gstvideoscale.c:
1395 * gst/videoscale/vs_image.c:
1396 * gst/videotestsrc/gstvideotestsrc.c:
1397 gst: Remove dead assignments and resulting unused variables
1399 2009-08-07 13:05:42 +0200 Josep Torra <n770galaxy@gmail.com>
1401 * docs/design/draft-va.txt:
1402 docs: add draft for generic introduction of video acceleration APIs idea
1404 2009-08-07 08:53:44 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
1406 * ext/theora/gsttheoradec.h:
1407 * ext/theora/theoradec.c:
1408 Revert "theora: Convert theoradec to libtheora 1.0 API"
1409 This reverts commit f1e142ac9dcfb754d85357b9077d5aee48559dd9.
1410 Temporarily revert until we have a workaround for debian/ubuntu
1411 packaging failure (see http://bugs.debian.org/528710).
1413 2009-08-07 09:32:00 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
1415 * gst/typefind/gsttypefindfunctions.c:
1416 typefindfunctions: Add typefinders for many game sound console formats supported by gme
1417 These are AY, GBS, GYM, KSS, SAP and VGM. SPC and NSF already had typefinders.
1419 2009-07-16 11:29:20 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
1421 * ext/ogg/gstoggmux.c:
1422 oggmux: fix warning when we're not linked downstream and error out properly
1423 Fix caps warning when there's no element linked downstream, and pass
1424 not-linked flow return value correctly up the chain, so we error out
1425 correctly. Fixes #588717.
1427 2009-07-31 14:59:03 -0700 David Schleef <ds@schleef.org>
1429 * ext/theora/gsttheoradec.h:
1430 * ext/theora/theoradec.c:
1431 theora: Convert theoradec to libtheora 1.0 API
1433 2009-08-06 20:47:33 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
1435 * ext/pango/gsttextrender.c:
1436 textrender: Fix blitting of text over the output buffer and cairo painting
1438 2009-08-06 09:13:14 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
1440 * ext/pango/gsttextrender.c:
1441 textrender: Fix endianness problems (i.e. make it work again on big endian architectures)
1443 2009-07-31 14:27:28 +0300 Stefan Kost <ensonic@users.sf.net>
1445 * tests/icles/test-colorkey.c:
1446 colorkey-test: fix xsync error
1448 2009-07-06 23:06:50 +0300 Siarhei Siamashka <siarhei.siamashka@nokia.com>
1450 * gst/ffmpegcolorspace/imgconvert.c:
1451 * gst/ffmpegcolorspace/imgconvert_template.h:
1452 ffmpegcolorspace: support for direct conversion from uyvy422 to rgb formats
1454 2009-07-14 12:33:29 +0300 Stefan Kost <ensonic@users.sf.net>
1456 * gst/playback/gstplaysink.c:
1457 playbin2: smarter sink selection. Fixes #588523
1458 Don't do fallbacks if application specified a sink element. When doing the
1459 fallback use configured default elements instead of hardcoded linux only
1460 elements. Improve error messages accordingly.
1462 2009-08-06 12:18:36 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
1464 * gst/playback/gstqueue2.c:
1465 queue2: post error message when pausing task if so appropriate
1466 If a downstream element returns an error while upstream has already
1467 put all data into queue2 (including EOS), upstream will no longer
1468 chain into queue2, so it is up to queue2 to perform some
1469 EOS handling / message posting in such cases. See #589991.
1471 2009-08-06 12:58:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1473 * gst-libs/gst/audio/gstbaseaudiosrc.c:
1474 baseaudiosrc: change default slave method
1475 Set the default slave method to the much better skew slaving algortihm.
1477 2009-08-06 12:01:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1479 * ext/pango/gsttextoverlay.c:
1480 textoverlay: make buffer writable
1481 Make the input buffer writable before changing its contents.
1483 2009-08-06 09:55:42 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
1485 * gst/typefind/gsttypefindfunctions.c:
1486 typefinding: fix postscript typefinder probability
1487 Two bytes for a rare format hardly warrants MAXIMUM typefinding
1488 probability, POSSIBLE seems more appropriate.
1490 2009-08-04 14:55:06 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
1492 * ext/pango/gsttextoverlay.c:
1493 pango: Send queries from the srcpad directly to the video sinkpad
1495 2009-08-04 14:32:51 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
1497 * gst/subparse/gstsubparse.c:
1498 subparse: Implement POSITION query
1500 2009-08-04 14:29:41 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
1502 * gst/subparse/gstsubparse.c:
1503 * gst/subparse/samiparse.c:
1504 subparse: Implement SEEKING query
1506 2009-08-04 14:14:53 +0200 John Millikin <jmillikin@gmail.com>
1509 * gst-libs/gst/tag/gstid3tag.c:
1510 * gst-libs/gst/tag/gstvorbistag.c:
1511 tag: Add support for ALBUM_ARTIST tag in vorbiscomments and ID3v2 tags
1512 Require latest core for this.
1515 2009-08-04 12:46:57 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
1517 * ext/pango/gsttextoverlay.c:
1518 * ext/pango/gsttextoverlay.h:
1519 pango: Add support for xRGB and BGRx formats
1521 2009-08-04 12:22:14 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
1523 * ext/pango/gsttextoverlay.c:
1524 pango: Fix endianness issues from the pangocairo switch
1525 cairo's ARGB is in native endianness, i.e. ARGB on big endian architectures
1526 and BGRA on little endian architectures.
1528 2009-08-04 12:11:00 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
1530 * ext/pango/gsttextoverlay.c:
1531 pango: Re-add shading support which was dropped by a previous patch
1533 2009-08-04 11:58:45 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
1536 * ext/pango/gsttextoverlay.c:
1537 pango: Check if pangocairo supports vertical rendering and fix properties
1539 2009-08-04 11:45:01 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
1541 * ext/pango/gsttextrender.c:
1542 textrender: Use PROP_X instead of ARG_X consistently
1544 2009-08-04 11:42:28 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
1546 * ext/pango/gstclockoverlay.c:
1547 * ext/pango/gsttextoverlay.c:
1548 * ext/pango/gsttextrender.c:
1549 * ext/pango/gsttimeoverlay.c:
1550 pango: Some minor cleanup
1552 2009-08-04 11:36:58 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
1555 pango: Check for pangocairo instead of pangoft2
1557 2009-08-04 11:35:10 +0200 Young-Ho Cha <ganadist@chollian.net>
1559 * ext/pango/gsttextoverlay.c:
1560 * ext/pango/gsttextoverlay.h:
1561 * ext/pango/gsttextrender.c:
1562 * ext/pango/gsttextrender.h:
1563 pango: Use pango-cairo instead of pango-ft2
1564 pango-cairo will always use the native font rendering backend
1565 of the platform and provides better results.
1568 2009-08-04 10:35:34 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
1570 * gst/typefind/gsttypefindfunctions.c:
1571 typefindfunctions: Add SVG typefinder
1573 2009-08-04 10:29:48 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
1575 * gst/typefind/gsttypefindfunctions.c:
1576 typefindfunctions: Add postscript typefinder
1578 2009-07-30 15:08:35 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
1580 * gst/typefind/gsttypefindfunctions.c:
1581 typefindfunctions: Use static caps again for MPEG4 typefinding
1583 2009-07-30 15:05:28 +0200 Arnout Vandecappelle <arnout@mind.be>
1585 * gst/typefind/gsttypefindfunctions.c:
1586 typefindfunctions: Implement better & more flexible MPEG4 typefinding
1587 This detects more MPEG4 streams as MPEG4.
1590 2009-07-30 14:04:30 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
1592 * gst-libs/gst/cdda/gstcddabasesrc.c:
1593 cddabasesrc: Allow to specify the device name in the URI
1594 The allowed URI scheme is now:
1595 cdda://(device#)?track
1596 Also allow every combination of uppercase and lowercase
1597 characters for the protocol part.
1600 2009-07-30 12:37:07 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
1602 * gst/videoscale/gstvideoscale.c:
1603 videoscale: Restrict width/height to 2^15 - 1
1604 Otherwise integer overflows will happen, resulting in segmentation faults.
1607 2009-07-29 14:55:04 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
1609 * gst/ffmpegcolorspace/imgconvert_template.h:
1610 ffmpegcolorspace: Fix indention of template header
1612 2009-07-29 14:10:35 +0200 Philip Jägenstedt <philipj@opera.com>
1614 * gst-libs/gst/app/gstappsrc.c:
1615 appsrc: Clarify documentation about caps and linkage
1618 2009-07-29 07:42:05 +0200 Benjamin Gaignard <benjamin@gaignard.net>
1620 * gst/typefind/gsttypefindfunctions.c:
1621 typefindfunctions: Fix typefinding of SDP files
1624 2009-07-28 20:50:06 +0200 Kipp Cannon <kcannon@ligo.caltech.edu>
1626 * gst/audioresample/gstaudioresample.c:
1627 audioresample: Take the output offsets from the input if possible
1630 2009-07-28 15:54:14 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
1632 * gst/videoscale/gstvideoscale.c:
1633 videoscale: Make sure to allocate enough memory for the temporary buffer
1634 and fix scaling of odd-height interlaced video.
1636 2009-07-28 15:18:56 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
1638 * gst/videoscale/gstvideoscale.c:
1639 videoscale: Fix interlaced scaling for I420
1640 ...and some other minor mistakes in the previous change.
1642 2009-07-28 14:12:31 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
1644 * gst/ffmpegcolorspace/avcodec.h:
1645 * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
1646 * gst/ffmpegcolorspace/gstffmpegcodecmap.h:
1647 * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
1648 * gst/ffmpegcolorspace/gstffmpegcolorspace.h:
1649 * gst/ffmpegcolorspace/imgconvert.c:
1650 ffmpegcolorspace: Include interlacing information in the AVPicture
1651 This later allows to handle interlaced AVPicture different than
1652 progressive ones which is needed for horizontally subsampled YUV
1653 formats, see bug #589242.
1655 2009-07-28 13:55:30 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
1657 * gst/videoscale/gstvideoscale.c:
1658 * gst/videoscale/gstvideoscale.h:
1659 videoscale: Add support for interlaced content
1660 videoscale is not mixing content of two seperate fields anymore
1661 and does scaling on every field separately.
1664 2009-08-06 01:44:24 +0100 Jan Schmidt <thaytan@noraisin.net>
1667 back to development -> 0.10.24.1
1669 2009-08-05 02:03:44 +0100 Jan Schmidt <thaytan@noraisin.net>
1671 * gst-plugins-base.doap:
1672 Add 0.10.24 release to the doap file
1674 === release 0.10.24 ===
1676 2009-08-05 00:56:58 +0100 Jan Schmidt <thaytan@noraisin.net>
1682 * docs/plugins/gst-plugins-base-plugins.args:
1683 * docs/plugins/gst-plugins-base-plugins.hierarchy:
1684 * docs/plugins/gst-plugins-base-plugins.interfaces:
1685 * docs/plugins/gst-plugins-base-plugins.prerequisites:
1686 * docs/plugins/gst-plugins-base-plugins.signals:
1687 * docs/plugins/inspect/plugin-adder.xml:
1688 * docs/plugins/inspect/plugin-alsa.xml:
1689 * docs/plugins/inspect/plugin-app.xml:
1690 * docs/plugins/inspect/plugin-audioconvert.xml:
1691 * docs/plugins/inspect/plugin-audiorate.xml:
1692 * docs/plugins/inspect/plugin-audioresample.xml:
1693 * docs/plugins/inspect/plugin-audiotestsrc.xml:
1694 * docs/plugins/inspect/plugin-cdparanoia.xml:
1695 * docs/plugins/inspect/plugin-decodebin.xml:
1696 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
1697 * docs/plugins/inspect/plugin-gdp.xml:
1698 * docs/plugins/inspect/plugin-gio.xml:
1699 * docs/plugins/inspect/plugin-gnomevfs.xml:
1700 * docs/plugins/inspect/plugin-libvisual.xml:
1701 * docs/plugins/inspect/plugin-ogg.xml:
1702 * docs/plugins/inspect/plugin-pango.xml:
1703 * docs/plugins/inspect/plugin-playback.xml:
1704 * docs/plugins/inspect/plugin-queue2.xml:
1705 * docs/plugins/inspect/plugin-subparse.xml:
1706 * docs/plugins/inspect/plugin-tcp.xml:
1707 * docs/plugins/inspect/plugin-theora.xml:
1708 * docs/plugins/inspect/plugin-typefindfunctions.xml:
1709 * docs/plugins/inspect/plugin-uridecodebin.xml:
1710 * docs/plugins/inspect/plugin-video4linux.xml:
1711 * docs/plugins/inspect/plugin-videorate.xml:
1712 * docs/plugins/inspect/plugin-videoscale.xml:
1713 * docs/plugins/inspect/plugin-videotestsrc.xml:
1714 * docs/plugins/inspect/plugin-volume.xml:
1715 * docs/plugins/inspect/plugin-vorbis.xml:
1716 * docs/plugins/inspect/plugin-ximagesink.xml:
1717 * docs/plugins/inspect/plugin-xvimagesink.xml:
1720 2009-08-05 00:38:40 +0100 Jan Schmidt <thaytan@noraisin.net>
1755 2009-08-01 17:26:23 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
1757 * gst/typefind/gsttypefindfunctions.c:
1758 * tests/check/gst/typefindfunctions.c:
1759 typefinding: fix detection of fLaC id packet in broken flac-in-ogg
1760 There are flac-in-ogg files without the usual flac packet framing
1761 and these files just have a 4-byte fLaC ID packet as first packet.
1762 We need to recognise the type just from these four bytes if we
1763 want oggdemux to recognise these streams correctly.
1765 2009-07-30 14:40:50 +0100 Jan Schmidt <thaytan@noraisin.net>
1801 0.10.24.5 pre-release
1803 2009-07-29 14:15:53 -0400 Olivier Crête <olivier.crete@collabora.co.uk>
1805 * gst-libs/gst/audio/gstaudiofilter.c:
1806 audiofilter: Don't assert on slightly different caps
1807 Plugins should not assert on incompatible caps, caps negotiation will
1810 2009-07-30 13:42:21 +0300 Stefan Kost <ensonic@users.sf.net>
1812 * gst/adder/gstadder.c:
1813 adder: reset pending flush-stop flag in state_changed. (mostly) Fixes #590146.
1815 2009-07-30 09:28:20 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
1818 configure: bump Gtk+ requirement of GUI examples from 2.12 to 2.14
1819 The gio mount example needs GtkMountOperation, which is new in 2.14.
1821 2009-07-27 10:29:27 +0100 Balachandran C <balachandran_c@rediffmail.com>
1823 * ext/alsa/gstalsasrc.c:
1824 alsasrc: set alsasrc->handle back to NULL when closing device
1825 Fixes crashes in gst_alsa_find_device_name() when probing or
1826 reading the device-name property (e.g. when doing a dot-file
1827 dump). Fixes #589797.
1829 2009-07-24 19:26:40 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
1831 * gst/playback/gststreamselector.c:
1832 playbin: rename GType of stream selector pad to avoid clash with input-selector from -bad
1833 Rename the GType of the pads of playbin's internal stream selector
1834 element so they don't use the same type name as input-selector's
1835 pads. Fixes #589622.
1837 2009-07-24 13:39:55 +0100 Jan Schmidt <thaytan@noraisin.net>
1870 0.10.23.4 pre-release
1872 2009-07-24 13:46:15 +0100 Jan Schmidt <thaytan@noraisin.net>
1874 * tests/examples/v4l/.gitignore:
1875 ignores: Ignore v4l probing example binary
1877 2009-07-24 09:35:38 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
1879 * gst/typefind/gsttypefindfunctions.c:
1880 typefind: recognise Kate spu subtitles as well
1881 Recognise spu-subtitles, SUB and K-SPU as valid categories for
1882 Kate subtitles as well.
1884 2009-07-24 00:42:16 +0300 Stefan Kost <ensonic@users.sf.net>
1887 Automatic update of common submodule
1888 From fedaaee to 94f95e3
1890 2009-07-22 14:21:43 +0100 Christian Schaller <christian.schaller@collabora.co.uk>
1892 * gst-plugins-base.spec.in:
1893 Update spec file with latest changes
1895 2009-07-20 17:28:20 +0100 Jan Schmidt <thaytan@noraisin.net>
1928 * win32/common/_stdint.h:
1929 * win32/common/audio-enumtypes.c:
1930 * win32/common/config.h:
1931 * win32/common/gstrtsp-enumtypes.c:
1932 * win32/common/interfaces-enumtypes.c:
1933 * win32/common/video-enumtypes.c:
1934 0.10.23.3 pre-release
1936 2009-07-20 12:51:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1938 * gst/audiotestsrc/gstaudiotestsrc.c:
1939 audiotestsrc: call send_event directly
1940 We can't call gst_element_send_event() from a streaming thread as it gets the
1941 state lock. Instead call the send_event method directly until we have a nice API
1942 for this in basesrc.
1945 2009-07-03 04:42:24 -0400 Olivier Crête <olivier.crete@collabora.co.uk>
1947 * gst-libs/gst/audio/gstaudiosink.c:
1948 audiosink: Add stream-status messages
1951 2009-07-03 04:41:05 -0400 Olivier Crête <olivier.crete@collabora.co.uk>
1953 * gst-libs/gst/audio/gstaudiosrc.c:
1954 audiosrc: Add stream-status messages
1957 2009-07-20 10:53:11 +0200 Edward Hervey <bilboed@bilboed.com>
1959 * gst/adder/gstadder.c:
1960 gstadder: Don't forget to free pending events on flush/dispose.
1963 2009-07-12 10:08:12 +0200 Edward Hervey <bilboed@bilboed.com>
1965 * tests/check/elements/adder.c:
1966 tests/adder: Add stream consistency checking. Fixes #588748
1968 2009-07-12 10:07:34 +0200 Edward Hervey <bilboed@bilboed.com>
1970 * gst/audiotestsrc/gstaudiotestsrc.c:
1971 audiotestsrc: Make sure tags are properly serialized. Fixes #588746
1972 We do this by letting the basesrc base class handle the tags.
1974 2009-07-13 09:28:54 +0200 Edward Hervey <bilboed@bilboed.com>
1976 * gst/adder/gstadder.c:
1977 * gst/adder/gstadder.h:
1978 adder: Collect incoming tag events and send them after newsegment. Fixes #588747
1980 2009-07-16 09:32:46 +0200 Edward Hervey <bilboed@bilboed.com>
1982 * ext/vorbis/vorbisdec.c:
1983 vorbisdec: Check for empty tag strings. Fixes #588724
1985 2009-07-14 17:03:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
1987 * gst/playback/gstqueue2.c:
1988 queue2: fix leak and improve buffering
1989 Keep track of the max requested position and compare this to the write position
1990 in the temp file to get the current amount of buffered data.
1991 Fix memleak of all incomming buffers.
1994 2009-07-15 17:40:14 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
1996 * gst/playback/Makefile.am:
1997 * gst/playback/gstinputselector.c:
1998 * gst/playback/gstinputselector.h:
1999 * gst/playback/gstplay-marshal.list:
2000 * gst/playback/gstplaybin2.c:
2001 playbin2: use private copy of input-selector
2002 We shouldn't really depend on elements from -bad for stream
2003 selection in playbin2, so use a private copy of input-selector
2004 until the selector plugin is ready to be moved to -base or -good.
2007 2009-07-15 17:26:32 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
2009 * gst/playback/gstinputselector.c:
2010 * gst/playback/gstinputselector.h:
2011 playback: add private copy of the input-selector from gst-plugins-bad
2012 Not hooked up yet though. See #586356.
2014 2009-07-14 19:00:36 +0200 Filippo Argiolas <filippo.argiolas@gmail.com>
2016 * tests/examples/v4l/Makefile.am:
2017 examples: fix v4l probe example build
2020 2009-07-14 19:00:10 +0100 Jan Schmidt <thaytan@noraisin.net>
2054 0.10.23.2 pre-release
2056 2009-07-14 16:24:10 +0100 Jan Schmidt <thaytan@noraisin.net>
2060 Add Turkish translations
2062 2009-07-14 15:31:13 +0100 Jan Schmidt <thaytan@noraisin.net>
2064 * tests/check/elements/adder.c:
2065 adder: One more attempt to fix the adder test
2066 Give up and discard and recreate the alsasrc after checking it can
2067 be opened, due to some strange crash inside alsa when we don't.
2069 2009-07-14 15:06:41 +0100 Jan Schmidt <thaytan@noraisin.net>
2071 * tests/check/elements/adder.c:
2072 adder: Perform get_state() in the unit test
2073 Wait for the alsasrc to return to NULL after setting it to PAUSED for
2074 testing, otherwise it leads to segfaults later on.
2076 2009-07-14 14:39:32 +0100 Jan Schmidt <thaytan@noraisin.net>
2078 * tests/check/elements/adder.c:
2079 adder: Don't fail when alsasrc is unavailable
2080 Make the liveadder test succeed silently when it can't be completed
2081 either because alsasrc is unavailable, or because the device is
2084 2009-07-13 22:51:48 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
2086 * gst-libs/gst/pbutils/descriptions.c:
2087 * gst/typefind/gsttypefindfunctions.c:
2088 typefinding: use subtitle/x-kate for Kate subtitle streams and application/x-kate for the rest
2089 Differentiate subtitle streams and lyrics/cracktastic/complex streams via
2090 the category string in the headers. This seems like a useful distinction
2091 to make, and also seems more future-proof. See #525743.
2093 2009-02-21 13:18:10 +0000 Vincent Penquerc'h <ogg.k.ogg.k@googlemail.com>
2095 * ext/ogg/gstoggmux.c:
2096 oggmux: add Kate caps to the list of accepted types
2099 2009-07-13 21:56:46 +0300 Stefan Kost <ensonic@users.sf.net>
2101 * gst/playback/gsturidecodebin.c:
2102 uridecodebin: treat uri-schemas incasesensitive
2103 Treat uri-schemas incasesensitive. This is mandated in rfc2396 section 3.1.
2104 Fixes not showing buffering messages e.g. for HTTP://...
2106 2009-07-13 21:54:47 +0300 Stefan Kost <ensonic@users.sf.net>
2108 * gst-libs/gst/interfaces/navigation.c:
2109 navigation: simplify docs
2110 Make short-desc short - its used in the toc. Strip uneeded markup.
2112 2009-07-13 18:31:15 +0100 Jan Schmidt <thaytan@noraisin.net>
2114 * win32/common/libgstnetbuffer.def:
2115 * win32/common/libgstvideo.def:
2117 Remove methods from video base classes that have moved to -bad.
2118 Add gst_netaddress_to_string
2120 2009-07-13 17:56:58 +0100 Jan Schmidt <thaytan@noraisin.net>
2122 * tests/examples/gio/.gitignore:
2123 ignores: ignore the giosrc-mounting example binary
2125 2009-07-13 17:54:40 +0100 Jan Schmidt <thaytan@noraisin.net>
2127 * gst-libs/gst/interfaces/navigation.c:
2128 navigation: Add some partial documentation
2129 Add a general documentation blurb for the GstNavigation functionality.
2130 Still lacks some example code and detail on how to implement it.
2132 2009-07-13 17:52:39 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
2134 * gst-libs/gst/pbutils/descriptions.c:
2135 pbutils: add description for Siren codec and make two descriptions non-translatable
2137 2009-07-13 12:23:20 -0400 Olivier Crête <olivier.crete@collabora.co.uk>
2140 Automatic update of common submodule
2141 From 5845b63 to fedaaee
2143 2009-07-13 18:21:49 +0200 Elliott Sales de Andrade <quantum.analyst at gmail.com>
2145 * gst-libs/gst/riff/riff-ids.h:
2146 * gst-libs/gst/riff/riff-media.c:
2147 riff: add siren to the RIFF parser
2148 Add siren7 caps to the RIFF parser.
2150 2009-07-13 14:55:59 +0200 Filippo Argiolas <filippo.argiolas@gmail.com>
2153 * tests/examples/Makefile.am:
2154 * tests/examples/v4l/Makefile.am:
2155 * tests/examples/v4l/probe.c:
2156 v4lsrc: add a simple test case for device probing
2158 2009-07-03 11:38:01 +0200 Filippo Argiolas <filippo.argiolas@gmail.com>
2161 * sys/v4l/Makefile.am:
2162 * sys/v4l/gstv4lelement.c:
2163 v4lsrc: optional support for device probing with gudev
2164 Enumerate v4l devices using gudev if available.
2167 2009-07-10 23:24:36 +0100 Stefan Kost <ensonic@users.sf.net>
2169 * gst/adder/gstadder.c:
2170 adder: add since tags to docs
2172 2009-07-10 21:29:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
2174 * tests/examples/seek/seek.c:
2175 seek: don't automatically start pipeline in DB
2176 Keep the pipeline paused when we detect download buffering. The user has to
2177 manually start the pipeline for now because we can't estimate when the buffering
2178 will finish or when we have underrun.
2180 2009-07-10 21:01:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
2182 * gst/playback/gstqueue2.c:
2183 queue2: flush differently, avoiding deadlocks
2184 Don't flush the file by closing and opening it but instead use g_freopen. This
2185 avoids a deadlock in shutdown because we emit the temp-location property change
2186 with the wrong lock held.
2188 2009-07-10 20:25:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
2190 * tests/examples/seek/seek.c:
2191 seek: add a checkbox for progressive download
2193 2009-07-10 20:24:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
2195 * gst/playback/gsturidecodebin.c:
2196 uridecodebin: Fix template construction
2197 Fix the construction of the temporary filename construction as the application
2198 name can be NULL and we don't want a separator between the prgname and the
2201 2009-07-10 20:04:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
2203 * gst/playback/gstplay-enum.c:
2204 * gst/playback/gstplay-enum.h:
2205 * gst/playback/gstplaybin2.c:
2206 playbin2: add support for progressive download
2207 Add a new playbin2 flag (initially disabled) to enable progressive download
2208 buffering in uridecodebin.
2210 2009-07-10 19:59:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
2212 * gst/playback/gsturidecodebin.c:
2213 uridecodebin: add download property
2214 Add a download property that will attempt to configure queue2 into progressive
2216 Make sure we only enable download buffering for quicktime and flv formats.
2218 2009-07-10 19:49:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
2220 * gst/playback/gstqueue2.c:
2221 queue2: add temp-template property
2222 Add a new temp-template property so that queue2 can securely allocate a
2223 temporary filename. Deprecate the temp-location property for setting the
2224 location but still use it to notify the allocated temp file.
2226 2009-07-10 20:06:28 +0100 Stefan Kost <ensonic@users.sf.net>
2228 * gst/adder/gstadder.c:
2229 * gst/adder/gstadder.h:
2230 adder: add a caps-property to avoid to need to plug a capsfilter afterwards
2231 Adder can only handle one common format accross the pads. Thus one needed to add
2232 a capsfilter afterwards and manage the caps. Now one can simply set the caps on
2235 2009-07-10 18:59:05 +0100 Stefan Kost <ensonic@users.sf.net>
2237 * tests/check/elements/adder.c:
2238 adder: skip live-seek text if we have no audiosrc, add new test
2239 The seek-test needs a real audiosrc. Also add a test that checks that adder is
2240 reusable. Finaly handle warnings as warnings to fix a assertion.
2242 2009-07-10 19:16:35 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2244 * ext/gio/gstgiosink.c:
2245 gio: Also post a "not-mounted" message from giosink
2247 2009-07-10 17:15:48 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2249 * tests/examples/gio/giosrc-mounting.c:
2250 gio: Remove workaround for playbin2 bug in the sample application
2251 The playbin2 bug was #588078.
2253 2009-07-10 17:08:40 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2255 * gst/playback/gstplaybin2.c:
2256 playbin2: Make it possible for READY->PAUSED to succeed after it failed the first time
2257 If READY->PAUSED failed in the source element we would've swapped
2258 the current and next group already. To allow READY->PAUSED to succeed
2259 after the first failure we have to swap the current and next group
2260 back again. This also ensure that we're again in the same state
2261 as before the failed state change and not at the next group.
2262 This was especially a problem for playbin2 pipelines that use the
2263 new mounting support in giosrc as the source would fail for READY->PAUSED
2264 the first time, the application mounts the location and then tries
2265 to go READY->PAUSED again (and this time it would succeed).
2268 2009-07-10 11:42:51 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2271 * tests/examples/Makefile.am:
2272 * tests/examples/gio/Makefile.am:
2273 * tests/examples/gio/giosrc-mounting.c:
2274 gio: Add example application that shows how to handle the "not-mounted" message
2276 2009-07-10 11:24:57 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2279 gio: Remove the experimental status from the GIO plugin
2282 2009-07-10 11:24:05 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2284 * ext/gio/gstgiosink.c:
2285 * ext/gio/gstgiosrc.c:
2286 gio: Add documentation for the new "not-mounted" and "file-exists" messages
2288 2009-07-09 13:45:13 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2290 * ext/gio/gstgiobasesrc.c:
2291 gio: Make sure that we have the correct stream position when starting
2293 2009-07-08 17:24:19 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2295 * ext/gio/gstgiobasesink.c:
2296 gio: Make sure to flush the output stream if it shouldn't be closed
2297 Otherwise there might still be unwritten data after the element
2300 2009-07-08 17:19:29 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2302 * ext/gio/gstgiobasesink.c:
2303 * ext/gio/gstgiobasesink.h:
2304 * ext/gio/gstgiobasesrc.c:
2305 * ext/gio/gstgiobasesrc.h:
2306 * ext/gio/gstgiosink.c:
2307 * ext/gio/gstgiosrc.c:
2308 gio: Don't close the GIO streams for the giostream{src,sink} elements
2309 This makes it possible to do something useful with the streams
2310 after the element has stopped. Fixes bug #587896.
2312 2009-07-08 17:19:05 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2314 * tests/check/pipelines/gio.c:
2315 gio: Try to reuse the pipeline with the same stream objects
2317 2009-07-08 17:02:54 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2319 * ext/gio/gstgiobasesink.c:
2320 * ext/gio/gstgiobasesrc.c:
2321 gio: Improve the error message if a stream is already closed before usage
2323 2009-07-08 16:55:41 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2325 * ext/gio/gstgiosink.c:
2326 gio: Post a custom file-exists message on the bus if the file already exists
2327 An application can handle this message, remove the file in question
2328 and restart the pipeline again without showing an error.
2329 This fixes bug #529300.
2331 2009-07-08 16:54:56 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2333 * ext/gio/gstgiosrc.c:
2334 gio: Use OPEN_READ instead of NOT_FOUND if a location is not mounted
2336 2009-07-08 16:50:56 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2338 * ext/gio/gstgiosink.c:
2339 gio: Use OPEN_WRITE instead of OPEN_READ as error category in giosink
2341 2009-07-08 15:52:35 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2343 * ext/gio/gstgiosrc.c:
2344 gio: Post a custom "not-mounted" message on the bus
2345 This allows applications to mount the GFile if possible and restart
2346 the pipeline instead of simply giving an error.
2348 2009-07-08 15:08:32 +0200 Philip Jägenstedt <philipj@opera.com>
2350 * gst/audioconvert/gstchannelmix.c:
2351 audioconvert: Fix compilation when debugging is disabled
2354 2009-07-07 20:23:23 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2356 * ext/gio/gstgiobasesink.c:
2357 * ext/gio/gstgiobasesink.h:
2358 * ext/gio/gstgiobasesrc.h:
2359 * ext/gio/gstgiosink.c:
2360 * ext/gio/gstgiosink.h:
2361 * ext/gio/gstgiostreamsink.c:
2362 * ext/gio/gstgiostreamsink.h:
2363 gio: Add vfunc for requesting the stream for the sinks too
2365 2009-07-07 20:21:36 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2367 * ext/gio/gstgiobasesink.c:
2368 * ext/gio/gstgiobasesink.h:
2369 * ext/gio/gstgiobasesrc.c:
2370 * ext/gio/gstgiosink.c:
2371 * ext/gio/gstgiosrc.c:
2372 * ext/gio/gstgiostreamsink.c:
2373 * ext/gio/gstgiostreamsrc.c:
2374 gio: Some more random cleanup
2376 2009-07-07 20:20:58 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2379 * ext/gio/gstgiobasesink.c:
2380 * ext/gio/gstgiobasesrc.c:
2381 * ext/gio/gstgiobasesrc.h:
2382 * ext/gio/gstgiosink.c:
2383 * ext/gio/gstgiosrc.c:
2384 * ext/gio/gstgiosrc.h:
2385 * ext/gio/gstgiostreamsink.c:
2386 * ext/gio/gstgiostreamsrc.c:
2387 * ext/gio/gstgiostreamsrc.h:
2388 gio: Update my mail address and copyright
2390 2009-07-07 20:18:00 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2392 * ext/gio/gstgiobasesrc.c:
2393 * ext/gio/gstgiobasesrc.h:
2394 * ext/gio/gstgiosrc.c:
2395 * ext/gio/gstgiostreamsrc.c:
2396 * ext/gio/gstgiostreamsrc.h:
2397 gio: General clean up and simplification
2398 The GInputStreams are now requested by a vfunc from
2399 the subclasses instead of relying that the subclass
2400 sets it until it's needed.
2401 This might also fix bug #587896.
2403 2009-07-06 22:31:12 +0100 Stefan Kost <ensonic@users.sf.net>
2405 * gst/adder/gstadder.c:
2406 adder: keep sending newsegments after seeking
2407 Adder sends with timestamps from 0 upwards. After seeking we need to send
2408 new-segments to get correct positions-queries.
2410 2009-07-06 20:44:00 +0100 Stefan Kost <ensonic@users.sf.net>
2412 * tests/check/elements/adder.c:
2413 adder: make test more robust
2414 Add audioconverts to the live-seeking test to make it negotiate.
2416 2009-06-30 17:19:50 +0300 Stefan Kost <ensonic@users.sf.net>
2418 * sys/xvimage/xvimagesink.c:
2419 xvimagesink: use core performance log category
2421 2009-07-05 21:29:40 +0200 Edward Hervey <bilboed@bilboed.com>
2423 * gst/adder/gstadder.c:
2424 adder: Call set_flushing(TRUE) for flushing seeks *when* the streaming is stopped.
2425 This ensures that collectpads' cookie is properly updated so that when the streaming
2426 threads will restart and be checking for the flushing status of all pads there will
2427 be no inconsistent state.
2429 2009-07-05 18:01:38 +0200 Hans-Peter Nilsson <hp@gcc.gnu.org>
2431 * ext/pango/gstclockoverlay.c:
2432 pango: Call tzset() before localtime_r()
2433 POSIX and your local friendly ctime(3) manual entry says that localtime_r isn't
2434 required to set the state variables that define the current timezone. Indeed,
2435 glibc (at least 2.9) doesn't do this for subsequent calls. The effect is that
2436 if the system timezone is changed for a running program between two calls to
2437 gst_clock_overlay_render_time, it won't be noticed. For glibc, changing the
2438 timezone equals /etc/localtime being modified.
2441 2009-07-01 17:33:14 -0700 David Schleef <ds@schleef.org>
2444 build: remove spurious schroedinger reference
2446 2009-07-01 10:25:43 -0700 David Schleef <ds@schleef.org>
2450 * ext/schroedinger/Makefile.am:
2451 * ext/schroedinger/gstschro.c:
2452 * ext/schroedinger/gstschrodec.c:
2453 * ext/schroedinger/gstschroenc.c:
2454 * ext/schroedinger/gstschroparse.c:
2455 * ext/schroedinger/gstschroutils.c:
2456 * ext/schroedinger/gstschroutils.h:
2457 * gst-libs/gst/video/Makefile.am:
2458 * gst-libs/gst/video/gstbasevideocodec.c:
2459 * gst-libs/gst/video/gstbasevideocodec.h:
2460 * gst-libs/gst/video/gstbasevideodecoder.c:
2461 * gst-libs/gst/video/gstbasevideodecoder.h:
2462 * gst-libs/gst/video/gstbasevideoencoder.c:
2463 * gst-libs/gst/video/gstbasevideoencoder.h:
2464 * gst-libs/gst/video/gstbasevideoparse.c:
2465 * gst-libs/gst/video/gstbasevideoparse.h:
2466 * gst-libs/gst/video/gstbasevideoutils.c:
2467 * gst-libs/gst/video/gstbasevideoutils.h:
2468 basevideo: send basevideo back to remedial school
2469 Move basevideo classes and schroedinger plugin to -bad.
2471 2009-07-01 12:54:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2473 * docs/libs/gst-plugins-base-libs-sections.txt:
2474 * gst-libs/gst/netbuffer/gstnetbuffer.h:
2475 netaddress: add constant for max len
2477 2009-07-01 12:48:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2479 * docs/libs/gst-plugins-base-libs-sections.txt:
2480 * gst-libs/gst/netbuffer/gstnetbuffer.c:
2481 * gst-libs/gst/netbuffer/gstnetbuffer.h:
2482 netbuffer: add gst_netaddress_to_string
2483 Add function to serialize a net address to a string.
2484 API: GstNetAddress::gst_netaddress_to_string()
2486 2009-06-30 18:44:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2488 * gst/playback/gsturidecodebin.c:
2489 uridecodebin: make fd:// uri use buffering too
2490 fd:// usually operate in push mode only and are thus suitable for buffering.
2492 2009-06-30 14:46:38 +0300 Stefan Kost <ensonic@users.sf.net>
2494 * gst/playback/gstplaybin2.c:
2495 * gst/volume/gstvolume.c:
2496 volume: include "1.0=100%" in property description
2498 2009-06-30 14:45:51 +0300 Stefan Kost <ensonic@users.sf.net>
2500 * gst/playback/gstplaysink.c:
2501 playsink: remove unused property defs
2503 2009-06-29 17:11:50 +0300 Stefan Kost <ensonic@users.sf.net>
2505 * gst-libs/gst/audio/multichannel.c:
2506 multichannel: rewrite the new doc comment a bit
2507 Its part of the audio lib.
2509 2009-06-29 14:34:02 +0100 Jan Schmidt <thaytan@noraisin.net>
2511 * gst/playback/gstplaysink.c:
2512 playsink: Avoid a segfault when the video sink fails to start
2513 Don't attempt to display the subpictures and segfault when the
2514 video sink failed to start (and hence the videochain is NULL).
2516 2009-06-29 15:14:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2518 * gst-libs/gst/audio/gstringbuffer.c:
2519 * gst-libs/gst/audio/gstringbuffer.h:
2520 ringbuffer: add vmethod to clear the ringbuffer
2521 Add a vmethod so that subclasses can be notified when they should clear the data
2524 2009-06-29 14:00:14 +0100 Jan Schmidt <thaytan@noraisin.net>
2526 * gst-libs/gst/riff/riff-media.c:
2527 riff-media: Fix the fourcc caps property for VC-1/WMVA
2528 The caps property for carrying fourccs is 'format', not 'fourcc'
2530 2009-06-29 12:20:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2532 * gst-libs/gst/rtsp/gstrtspconnection.c:
2533 rtsp: include in.h for FreeBSD compat
2536 2009-06-29 12:20:20 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2538 * win32/common/libgstapp.def:
2539 defs: add defs for new appsink buffer-list method
2541 2009-06-29 12:14:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2543 * gst-libs/gst/app/gstappsink.c:
2544 * gst-libs/gst/app/gstappsink.h:
2545 appsink: add docs and signals
2546 Add docs for the new callback.
2547 Add signals for the new buffer-list support.
2549 2009-06-29 10:24:36 +0200 Branko Subasic <branko@lnxbranko2.se.axis.com>
2551 * tests/check/elements/appsink.c:
2552 Added unit tests for buffer list support in appsink.
2554 2009-06-17 11:12:08 +0200 Branko Subasic <branko@lnxbranko2.se.axis.com>
2556 * gst-libs/gst/app/gstappsink.c:
2557 Added buffer list support.
2559 2009-06-17 09:23:11 +0200 Branko Subasic <branko@lnxbranko2.se.axis.com>
2561 * gst-libs/gst/app/gstappsink.h:
2562 Added buffer list support.
2564 2009-06-29 09:36:27 +0200 Peter Kjellerstedt <pkj@axis.com>
2566 * gst-libs/gst/sdp/gstsdpmessage.c:
2567 sdp: Include winsock2.h after defining WINVER.
2568 Similar to bug #587080.
2570 2009-06-29 09:31:40 +0200 Peter Kjellerstedt <pkj@axis.com>
2572 * gst-libs/gst/rtsp/gstrtspconnection.c:
2573 rtsp: Moved a comment.
2575 2009-06-27 23:23:02 +0300 Stefan Kost <ensonic@users.sf.net>
2577 * gst-libs/gst/audio/audio.c:
2578 * gst-libs/gst/audio/multichannel.c:
2579 docs: add basic section docs for multichannel and relocate the ones for audio
2580 Add section docs for multichannel, so that it has a short desc in the toc too.
2581 Move the section docs in adio up, so that the follow the copyright like
2584 2009-06-26 21:11:45 +0300 Stefan Kost <ensonic@users.sf.net>
2586 * sys/v4l/gstv4lelement.c:
2587 * sys/v4l/gstv4lsrc.c:
2588 v4l: open/close device in ready.
2589 Simillar change like in v4l2src. This allows probing feature in paused, where
2590 streaming is noit yet started.
2592 2009-06-10 17:05:22 +0300 René Stadler <rene.stadler@nokia.com>
2594 * gst/playback/gstplaysink.c:
2595 playbin2: fix initial volume handling also when reusing the element
2596 This is a follow-up to commit 452988, making it work correctly when the audio
2599 2009-06-26 21:48:58 +0400 Руслан Ижбулатов <lrn1986@gmail.com>
2601 * gst-libs/gst/rtsp/gstrtspconnection.c:
2602 Define WINVER before including any win headers
2605 2009-06-27 00:50:54 +0300 René Stadler <mail@renestadler.de>
2607 * gst-libs/gst/riff/riff-read.c:
2608 riff: prevent crash if rounded up tag size exceeds data size
2609 When rounding up `tsize' exceeds the remaining buffer size, `size' underflows
2610 and an invalid read past the buffer data follows.
2612 2009-06-26 15:17:21 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2614 * gst-libs/gst/video/gstbasevideocodec.c:
2615 basevideocodec: By default don't allow caps changes on the srcpad
2616 This fixed playback of Dirac files with schrodec when upstream wants
2617 a different width/height, basevideocodec accepts this and then
2618 pushes buffers with new caps but content of the old caps.
2619 In the best case this will just result in wrong unit size and a
2620 failure in basestransform elements.
2622 2009-06-26 14:11:21 +0100 Jan Schmidt <thaytan@noraisin.net>
2625 autogen.sh: Use printf instead of 'echo -n'. Check for automake-1.1[01]
2626 Check for more automake command variants. Use printf instead of 'echo -n'
2629 2009-06-26 13:41:38 +0100 Jan Schmidt <thaytan@noraisin.net>
2632 Automatic update of common submodule
2633 From f810030 to 5845b63
2635 2009-06-26 13:14:02 +0300 Stefan Kost <ensonic@users.sf.net>
2637 * gst/playback/gstscreenshot.c:
2638 screenshot: don't leak message
2640 2009-06-25 12:04:59 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
2642 * gst/typefind/gsttypefindfunctions.c:
2643 typefinding: lower the h264 typefinder's probability
2644 A NEARLY_CERTAIN is absolutely not warranted given the kind
2645 of things it checks for. Even a LIKELY is probably not entirely
2648 2009-06-24 15:13:56 +0100 Jan Schmidt <jan.schmidt@sun.com>
2651 Automatic update of common submodule
2652 From f3bb51b to f810030
2654 2009-06-24 09:48:41 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
2656 * gst-libs/gst/pbutils/descriptions.c:
2657 pbutils: add description for multipart
2658 So we get slightly nicer error messages when multipartdemux is missing.
2660 2009-06-23 18:07:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2662 * gst/adder/gstadder.c:
2663 adder: only unflush when we flushed before
2664 Ass suggested by Stefan Kost:
2665 Keep track of when the sinkpad was set to flushing and unflush the pad when an
2666 upstream flushing seek failed.
2668 2009-06-23 15:10:37 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
2670 * gst/playback/gsturidecodebin.c:
2671 uridecodebin: fix leak when the source fails to change state
2673 2009-06-23 12:40:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2675 * gst/subparse/gstssaparse.c:
2676 ssaparse: avoid leaking all buffers
2678 2009-06-22 22:18:03 +0300 Stefan Kost <ensonic@users.sf.net>
2680 * tests/check/elements/adder.c:
2681 adder: test seek handling in adder
2682 This tests seeking on an adder that has a normal and a live source connected.
2683 Wheter the current behavior is the desired one needs to be discussed still
2686 2009-06-22 16:17:10 +0300 Stefan Kost <ensonic@users.sf.net>
2688 * sys/ximage/ximagesink.c:
2689 * sys/xvimage/xvimagesink.c:
2690 x(v)imagesink: pass the xwindow along to not look at the yet unset var.
2691 When we call this from xwindow_new, x(v)imagesink->xwindow is not yet set.
2693 2009-06-22 11:40:33 +0300 Stefan Kost <ensonic@users.sf.net>
2695 * sys/ximage/ximagesink.c:
2696 * sys/ximage/ximagesink.h:
2697 * sys/xvimage/xvimagesink.c:
2698 * sys/xvimage/xvimagesink.h:
2699 x(v)imagesink: catch tags and show title in own window
2700 Refactor the code that sets the window title. Catch tag-events and use title
2701 metadata for the window title.
2703 2009-06-21 19:42:15 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2705 * gst/audiotestsrc/gstaudiotestsrc.c:
2706 audiotestsrc: Name gaussian noise "gaussian-noise" instead of just "gaussian"
2707 Also make all the function arrays constant.
2709 2009-06-21 12:27:37 +0200 Kipp Cannon <kcannon@ligo.caltech.edu>
2711 * gst/audiotestsrc/gstaudiotestsrc.c:
2712 * gst/audiotestsrc/gstaudiotestsrc.h:
2713 audiotestsrc: Add support for generating gaussian white noise
2714 This patch adds support for stationary white Gaussian noise.
2715 The Box-Muller algorithm is used to generate pairs of independent
2716 normally-distributed random numbers.
2719 2009-06-20 23:46:28 +0100 Jan Schmidt <thaytan@noraisin.net>
2721 * gst/ffmpegcolorspace/imgconvert.c:
2722 * gst/ffmpegcolorspace/imgconvert_template.h:
2723 ffmpegcolorspace: Fix NV12 and NV21 transformations
2724 Fix some stride problems, fix the nv12 to nv21 direct transformation,
2725 and implement a direct conversion to yuv444 to save CPU.
2727 2009-06-20 22:36:21 +0100 Jan Schmidt <thaytan@noraisin.net>
2729 * gst/videotestsrc/videotestsrc.c:
2730 videotestsrc: Fix NV12 painting for odd strides/heights
2732 2009-06-19 22:16:43 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
2734 * ext/cdparanoia/gstcdparanoiasrc.c:
2735 cdparanoia: run-time license is LGPL now that we require cdparanoia 0.10.2
2736 cdparanoia has an LGPL v2.1 license since 0.10.1 and we now require 0.10.2.
2737 Finally fixes #531035.
2739 2009-06-19 21:25:54 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
2741 * ext/cdparanoia/gstcdparanoiasrc.c:
2742 cdparanoia: try to guess a good cache size if it's set to -1
2743 Try to guess from the paranoia-mode setting whether playback or
2744 ripping is wanted, and use a smaller cache size if we're likely
2745 to be doing playback, to avoid a long startup delay. Since this
2746 was the value used in older cdparanoia versions, it should be
2747 fine in any case. See #586331.
2749 2009-06-19 11:27:40 +1000 Jonathan Matthew <jonathan@d14n.org>
2752 * ext/cdparanoia/gstcdparanoiasrc.c:
2753 * ext/cdparanoia/gstcdparanoiasrc.h:
2754 cdparanoia: expose cache size setting
2755 This setting was added in cdparanoia 10.2. The default value is good
2756 for audio extraction, but lower values (previous versions of cdparanoia
2757 used 150) are better for realtime playback.
2760 2009-06-19 17:43:03 +0100 Christian Schaller <christian.schaller@collabora.co.uk>
2762 * gst-plugins-base.spec.in:
2763 Make build of schro plugin conditional
2765 2009-06-19 15:52:34 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2767 * docs/libs/gst-plugins-base-libs-sections.txt:
2768 * gst-libs/gst/rtp/gstbasertppayload.c:
2769 * gst-libs/gst/rtp/gstbasertppayload.h:
2770 * win32/common/libgstrtp.def:
2771 basertppayload: add support for bufferlists
2772 Based on patch from Ognyan Tonchev.
2775 2009-06-19 15:33:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2777 * gst-libs/gst/rtp/gstrtpbuffer.c:
2778 rtpbuffer: use new convenience functions
2779 New core convenience functions makes the list getters and setters trivial.
2780 Maybe even too trivial...
2782 2009-06-18 19:07:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2784 * win32/common/libgstrtp.def:
2785 defs: add new symbol to win32 defs file
2786 Based on patches by Ognyan Tonchev.
2789 2009-06-18 19:04:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2791 * docs/libs/gst-plugins-base-libs-sections.txt:
2792 * gst-libs/gst/rtp/gstrtpbuffer.c:
2793 rtp: cleanups, add _list_get_seq() too
2794 Clean up the docs a little.
2795 Add missing _list_get_seq method.
2796 Add new symbols to the docs
2798 2009-06-18 18:47:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2800 * gst-libs/gst/rtp/gstrtpbuffer.c:
2801 * win32/common/libgstrtp.def:
2803 Add Since tags to docs
2804 Move some code around
2807 2009-06-18 17:46:01 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2809 * gst-libs/gst/rtp/gstrtpbuffer.c:
2810 * gst-libs/gst/rtp/gstrtpbuffer.h:
2811 * tests/check/libs/rtp.c:
2812 rtp: add bufferlist support
2814 2009-06-18 18:03:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2816 * gst-libs/gst/rtp/gstrtpbuffer.c:
2817 rtp: pass data to macros instead of GstBuffer
2819 2009-06-18 17:42:10 +0100 Jan Schmidt <thaytan@noraisin.net>
2821 * win32/common/libgstrtsp.def:
2822 win32: Add gst_rtsp_watch_queue_data() to the exports
2823 Fix the tests by exporting the new symbol from the win32 dlls
2825 2009-06-18 18:13:22 +0300 Stefan Kost <ensonic@users.sf.net>
2827 * sys/xvimage/xvimagesink.c:
2828 xvimagesink: appname might be NULL
2829 Don't set title if appname is unknown.
2831 2009-06-18 17:58:06 +0300 Stefan Kost <ensonic@users.sf.net>
2833 * sys/xvimage/xvimagesink.c:
2834 xvimagesink: set window title from application name
2836 2009-06-09 19:14:00 +0200 Peter Kjellerstedt <pkj@axis.com>
2838 * gst-libs/gst/rtsp/gstrtspurl.c:
2839 rtsp: Made the parsing of the RTSP URL scheme more generic.
2841 2009-06-15 13:58:26 +0200 Peter Kjellerstedt <pkj@axis.com>
2843 * gst-libs/gst/rtsp/gstrtspconnection.c:
2844 * gst-libs/gst/rtsp/gstrtspconnection.h:
2845 rtsp: Added gst_rtsp_watch_queue_data().
2846 gst_rtsp_watch_queue_data() is similar to gst_rtsp_watch_queue_message()
2847 but allows for queuing any data block for writing (much like
2848 gst_rtsp_connection_write() vs. gst_rtsp_connection_send().)
2849 API: gst_rtsp_watch_queue_data()
2851 2009-06-09 16:37:09 +0200 Peter Kjellerstedt <pkj@axis.com>
2853 * gst-libs/gst/rtsp/gstrtspconnection.c:
2854 rtsp: Only extract the session ID from RTSP responses.
2856 2009-06-09 19:06:57 +0200 Peter Kjellerstedt <pkj@axis.com>
2858 * gst-libs/gst/rtsp/gstrtspurl.c:
2859 rtsp: Added support for parsing IPv6 addresses in RTSP URLs.
2861 2009-06-09 14:31:18 +0200 Peter Kjellerstedt <pkj@axis.com>
2863 * gst-libs/gst/rtsp/gstrtspconnection.c:
2864 rtsp: Use getaddrinfo() to support both IPv4 and IPv6.
2866 2009-06-17 15:37:53 +0200 Peter Kjellerstedt <pkj@axis.com>
2868 * gst-libs/gst/rtsp/gstrtspconnection.c:
2869 rtsp: Improved base64 decoding in fill_bytes().
2870 The base64 decoding in fill_bytes() expected the size of the read data to
2871 be evenly divisible by four (which is true for the base64 encoded data
2872 itself). This did not, however, take whitespace (especially line breaks)
2873 into account and would fail the decoding if any whitespace was present.
2875 2009-06-17 14:00:23 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2877 * gst-libs/gst/audio/gstbaseaudiosrc.c:
2878 audiosrc: fix get_offset
2879 When we need to jump to the most recently captured sample, jump to where the
2880 next sample will be written instead of to some old data.
2883 2009-06-17 13:18:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2885 * gst-libs/gst/audio/gstbaseaudiosink.c:
2886 audiosink: free the ringbuffer when going to NULL
2887 Unparent and free the ringbuffer when going to NULL, like we do with the
2888 audiosrc element. We can do this now because we correctly manage the time
2891 2009-06-17 13:17:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2893 * gst-libs/gst/audio/gstaudiosink.c:
2894 * gst-libs/gst/audio/gstaudiosrc.c:
2895 audio: correctly handle short read/writes
2897 2009-05-05 15:37:54 +0300 René Stadler <rene.stadler@nokia.com>
2899 * gst-libs/gst/audio/gstbaseaudiosrc.c:
2900 baseaudiosrc: add some extra logging for buffer timestamps
2902 2009-06-17 11:22:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2904 * gst/adder/gstadder.c:
2905 adder: more seeking fixes.
2906 When a seek failed upstream, make sure the adder sinkpad is set unflushing again
2907 so that streaming can continue.
2908 We only have a pending segment when we flushed.
2909 Set the flush_stop_pending flag inside the appropriate locks and before we
2910 attempt to perform the upstream seek.
2911 Add some more comments.
2912 Use the right lock to protect the flags in flush_stop.
2915 2009-06-17 07:24:53 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2917 * gst/playback/gstdecodebin2.c:
2918 decodebin2: Free iterator after removing all groups
2920 2009-06-16 19:38:17 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2922 * gst-libs/gst/video/gstvideofilter.c:
2923 videofilter: Add a default get_unit_size function
2924 This returns the correct values for all formats that are handled by
2925 GstVideoFormat and makes all the custom get_unit_size functions in
2926 many elements unnecessary.
2928 2009-06-16 18:57:20 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2930 * gst-libs/gst/rtsp/gstrtspdefs.c:
2931 * gst-libs/gst/rtsp/gstrtspdefs.h:
2932 rtsp: add Timestamp header field
2935 2009-06-16 18:15:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2937 * gst/playback/gstplaybin2.c:
2938 playbin2: set smarter target state on uridecodebin
2939 Set the target state of the newly added uridecodebins to somthing else that
2940 PAUSED so that we keep their state in sync with the playsink state.
2943 2009-06-16 18:13:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2945 * gst/playback/gstplaysink.c:
2946 playsink: set the sink flag on the element
2948 2009-06-16 18:09:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2950 * gst/playback/gsturidecodebin.c:
2951 uridecodebin: add debug message
2953 2009-06-16 14:05:04 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
2955 * gst-libs/gst/audio/gstaudiosink.c:
2956 * gst-libs/gst/audio/gstaudiosrc.c:
2957 audiosink, audiosrc: do the class_ref()s in the right class_init functions
2958 Spotted by Philip Jägenstedt. Hopefully fixes #585970 for real.
2960 2009-06-15 15:39:09 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
2962 * gst-libs/gst/audio/gstaudiosink.c:
2963 * gst-libs/gst/audio/gstaudiosrc.c:
2964 audiosink,audiosrc: ref the audio ring buffer class and type in class_init
2965 Hack around thread-safety issues in GObject and our racy _get_type()
2966 functions (we could easily fix the _get_type() functions, but we still
2967 need to hack around the GObject class races until we require a newer
2968 GLib version, I think).
2970 2009-06-15 12:57:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
2972 * gst-libs/gst/audio/gstbaseaudiosrc.c:
2973 audiosrc: return FALSE when receiving a SEEK event
2974 When receiving a seek event, return FALSE as we don't implement seeking.
2976 2009-06-15 11:06:25 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
2978 * tests/examples/seek/seek.c:
2979 Don't use deprecated GTK API
2982 2009-06-15 11:40:00 +0300 Stefan Kost <ensonic@users.sf.net>
2984 * gst/adder/gstadder.c:
2985 adder: send flush_stop when seeking failed
2986 At least do the fix to sent the flush_stop when seeking failed to ensure we
2987 keep no pads flushing. before it was send when the seeking worked which is just
2988 plain wrong and was not the intention.
2990 2009-06-12 15:17:14 +0200 Peter Kjellerstedt <pkj@axis.com>
2992 * gst-libs/gst/rtsp/gstrtspconnection.c:
2993 rtsp: Use a more consistent naming of GstRTSPRec variables.
2995 2009-06-12 15:11:05 +0200 Peter Kjellerstedt <pkj@axis.com>
2997 * gst-libs/gst/rtsp/gstrtspconnection.c:
2998 * gst-libs/gst/rtsp/gstrtspconnection.h:
2999 rtsp: Call message_sent() callback for all sent messages.
3000 Previously the messages_sent() callback was only called for messages
3001 which had a CSeq, which excluded all data messages. Instead of using the
3002 CSeq as ID, use a simple index counter.
3004 2009-06-14 22:13:41 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
3006 * ext/ogg/gstoggdemux.c:
3007 * ext/theora/theoradec.c:
3008 * ext/vorbis/vorbisdec.c:
3009 oggdemux: post/send tags with the container-format tag
3010 For this to work properly, theoradec and vorbisdec need to put
3011 tag events received from upstream into the pending_events list
3012 so they get pushed out after any newsegment event, not before.
3014 2009-06-14 20:30:59 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3016 * tests/examples/seek/scrubby.c:
3017 * tests/examples/seek/seek.c:
3018 * tests/old/examples/seek/cdplayer.c:
3019 Don't use deprecated GTK API
3022 2009-06-12 16:31:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3024 * gst/adder/gstadder.c:
3025 adder: send flush-stop earlier
3026 When no flush-stop has been sent by upstream, we have to send one ourselves to
3027 continue playback. Do this as soon as the collect function is called instead of
3028 after we possibly pushed segment events (that got then flushed out)
3030 2009-06-12 13:55:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3032 * tests/examples/seek/seek.c:
3033 seek: add shuttle controls
3035 2009-06-12 13:55:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3037 * tests/examples/seek/stepping2.c:
3038 example: fix compile
3040 2009-06-12 13:52:25 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3042 * tests/examples/seek/Makefile.am:
3043 examples: build the stepping2 example
3045 2009-06-12 13:52:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3047 * gst/playback/gstplaysink.c:
3048 playsink: update for new step API
3050 2009-06-12 13:22:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3052 * ext/ogg/gstoggdemux.c:
3053 oggdemux: do reverse seeks more accurate
3054 For reverse seeking with the accurate flag set, try to be more precise by
3055 seeking a little bit after the requested position.
3057 2009-06-11 22:32:28 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
3059 * ext/ogg/gstogmparse.c:
3060 * gst/subparse/gstssaparse.c:
3061 * gst/subparse/gstssaparse.h:
3062 * gst/subparse/gstsubparse.c:
3063 * gst/subparse/gstsubparse.h:
3064 subparse, ogmparse: post tags with GST_TAG_SUBTITLE_CODEC
3065 Make subtitle parsers post a taglist with codec tags, so the application
3066 knows what kind of subtitle a subtitle stream is. Fixes #576552.
3068 2009-06-11 19:12:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3070 * gst-libs/gst/audio/gstringbuffer.c:
3071 ringbuffer: handle border cases in resampler
3073 2009-06-11 13:28:20 +0100 Jan Schmidt <thaytan@noraisin.net>
3076 * docs/libs/Makefile.am:
3077 * docs/plugins/Makefile.am:
3078 docs: Update common. Use upload-doc.mak instead of upload.mak
3080 2009-06-11 12:39:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3082 * gst-libs/gst/rtp/gstbasertppayload.c:
3085 2009-06-11 12:17:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3087 * gst-libs/gst/audio/gstbaseaudiosink.c:
3088 baseaudiosink: reset accum when dropping samples
3089 When we are resampling and we drop samples because we paused, reset the accum
3090 counter because it's now invalid.
3092 2009-06-11 11:16:15 +0100 Jan Schmidt <thaytan@noraisin.net>
3094 * docs/libs/gst-plugins-base-libs-sections.txt:
3095 * gst-libs/gst/interfaces/mixer.h:
3096 * gst-libs/gst/video/gstbasevideodecoder.h:
3097 docs: Fix a couple of warnings from the docs build.
3099 2009-06-10 21:36:19 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
3101 * gst-libs/gst/audio/testchannels.c:
3102 Don't include config.h multiple times when build audio testchannel app.
3103 Fixes build problem on win32 (#585075).
3105 2009-06-10 16:56:51 +0100 Jan Schmidt <thaytan@noraisin.net>
3107 * gst/playback/gstplaybin2.c:
3108 * gst/playback/gsturidecodebin.c:
3109 playbin2/uridecodebin: Fix connection-speed propagation
3110 uridecodebin expects the passed connection-speed value in kbps, so we
3111 need to divide the value stored in bps by 1000. Also, lower the upper
3112 limit on the properties to the value that we can actually store in our
3113 internal guint (which is plenty high enough)
3115 2009-06-10 14:37:36 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
3117 * gst/subparse/gstsubparse.c:
3118 * tests/check/elements/subparse.c:
3119 subparse: recognise more subrip timestamp variants
3120 Be even less restrictive in what we accept for .srt timestamps when
3121 typefinding and parsing subrip subtitles and add a unit test for
3122 the 'new' format. Fixes #585197.
3124 2009-06-09 22:00:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3126 * gst-libs/gst/rtsp/gstrtsptransport.h:
3127 rtsp: add some more docs
3129 2009-06-09 18:24:55 +0200 Peter Kjellerstedt <pkj@axis.com>
3131 * gst-libs/gst/rtsp/gstrtspmessage.c:
3132 rtsp: Avoid a compiler warning.
3134 2009-06-09 18:23:28 +0200 Peter Kjellerstedt <pkj@axis.com>
3136 * gst-libs/gst/rtsp/gstrtspdefs.h:
3137 rtsp: Updated documentation for GstRTSPResult.
3138 Moved GST_RTSP_ELAST to be last in the documentation to match the actual
3141 2009-05-20 17:30:23 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
3144 autogen: remove -Wno-portability from here
3145 as it is in configure.ac now.
3147 2009-06-09 16:28:20 +0200 Peter Kjellerstedt <pkj@axis.com>
3149 * gst-libs/gst/rtsp/gstrtspconnection.c:
3150 rtsp: Plug a memory leak.
3151 Free memory related to any partially read and/or written RTSP messages.
3153 2009-06-09 12:09:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3155 * gst-libs/gst/audio/gstbaseaudiosink.c:
3156 baseaudiosink: no need to cause discont when clipping
3157 Remove the discont-when-clipping hack now that basesink provides us with
3158 correctly clipped samples when stepping.
3160 2009-06-08 17:26:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3162 * gst-libs/gst/audio/gstbaseaudiosink.c:
3163 audiosink: don't align when we clip
3164 Don't align samples when they were clipped. Not entirely correct but better than
3167 2009-06-08 16:41:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3169 * tests/examples/seek/.gitignore:
3170 * tests/examples/seek/stepping2.c:
3171 examples: add stepping example in PLAYING
3172 Add stepping example in PLAYING, audio is a bit distorted because basesink does
3173 not provide good clipping info yet.
3175 2009-06-08 10:25:00 +0200 Edward Hervey <bilboed@bilboed.com>
3177 * gst-libs/gst/pbutils/descriptions.c:
3178 pbutils: Add description for hdv/aux-* formats.
3180 2009-06-07 22:20:33 +0400 LRN <lrn1986@gmail.com>
3182 * ext/schroedinger/Makefile.am:
3183 Added libgstbase to schro's LIBADD
3186 2009-06-06 02:15:05 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
3188 * gst-libs/gst/tag/gstid3tag.c:
3189 libgsttag: don't extract genres from empty ID3v1 tags
3190 If we don't have any other info, don't try to interpret the
3191 genre field. In particular we don't want to interpret a genre
3192 of 0 as 'Blues' if no other fields are set and the entire tag
3195 2009-06-05 18:13:25 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3197 * gst/playback/gstdecodebin2.c:
3198 decodebin2: make sure varargs are of right type
3199 Explicitly cast the variables to g_object_set to their right types.
3201 2009-06-05 16:49:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3203 * gst/playback/gstdecodebin2.c:
3204 decodebin2: increase stream probing queues
3205 When we are probing for streams, we want to set the queue size in such a way
3206 that we can scan a maximum amount of data without consuming too much memory.
3207 Therefore, remove the time limit on the queue and only stop scanning after 2MB
3211 2009-06-05 14:06:17 +0200 Peter Kjellerstedt <pkj@axis.com>
3213 * gst-libs/gst/rtsp/gstrtspconnection.c:
3216 2009-06-05 14:05:54 +0200 Peter Kjellerstedt <pkj@axis.com>
3218 * gst-libs/gst/rtsp/gstrtspconnection.c:
3219 rtsp: Remove an unused variable.
3221 2009-06-05 13:59:14 +0200 Peter Kjellerstedt <pkj@axis.com>
3223 * gst-libs/gst/rtsp/gstrtspconnection.c:
3224 rtsp: Removed duplicate initialization of conn->writefd.
3226 2009-06-05 13:55:08 +0200 Peter Kjellerstedt <pkj@axis.com>
3228 * gst-libs/gst/rtsp/gstrtspconnection.c:
3229 rtsp: Use #defined status codes.
3231 2009-06-05 13:53:29 +0200 Peter Kjellerstedt <pkj@axis.com>
3233 * gst-libs/gst/rtsp/gstrtspconnection.c:
3234 rtsp: Correct gen_tunnel_reply().
3235 Prevent gen_tunnel_reply() from generating an incomplete response
3236 in case an error response code is given.
3238 2009-06-05 10:57:44 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
3241 * win32/common/_stdint.h:
3242 * win32/common/config.h:
3243 * win32/common/video-enumtypes.c:
3244 configure: remove AC_C_INLINE which is not needed and causes problems with MSVC
3245 See #584835. Also update win32 files while we're at it.
3247 2009-06-04 08:57:24 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3249 * gst/playback/gstplaybin2.c:
3250 playbin2: API: Add {audio,video,text}-tags-changed signals
3253 2009-06-03 20:42:39 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
3255 * ext/vorbis/vorbisdec.c:
3256 vorbisdec: don't put invalid bitrate values into the taglist
3257 Bitrates are stored as 32-bit signed integers in the vorbis
3258 identification headers, but seem to be read incorrectly,
3259 namely as unsigned 32-bit integers, into the vorbis structure
3260 members which are of type long, which makes our check for
3261 values <= 0 fail with files that put -1 in there for unset
3264 2009-06-03 15:52:54 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3266 * tests/examples/seek/.gitignore:
3267 ignore: add new stepping app to ignore
3269 2009-06-03 15:31:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3271 * tests/examples/seek/Makefile.am:
3272 * tests/examples/seek/stepping.c:
3273 examples: add stepping example.
3274 Add an example of using playbin2 and frame stepping to simulate variable rate
3275 playback based on a sine wave.
3277 2009-06-03 12:45:08 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3279 * gst/playback/gstplaybin2.c:
3280 * gst/playback/gstplaysink.h:
3281 playbin2: also set custom text and subp sinks
3282 Set the custom subpicture and text sinks along with the custom audio and video
3284 Fix a little docs blurb too.
3286 2009-06-02 12:10:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3288 * gst-libs/gst/rtsp/gstrtspconnection.c:
3289 * gst-libs/gst/rtsp/gstrtspconnection.h:
3290 rtsp: add G_LIKELY because we can
3292 2009-06-02 09:53:05 +0200 Edward Hervey <bilboed@bilboed.com>
3294 * gst/typefind/gsttypefindfunctions.c:
3295 typefindfunctions: Fix caps for ogg typefinder.
3297 2009-05-29 11:10:07 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
3299 * docs/libs/gst-plugins-base-libs-sections.txt:
3300 docs: remove some cruft from -sections.txt file
3302 2009-06-01 11:31:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3304 * gst/playback/gstplaysink.c:
3305 * tests/examples/seek/seek.c:
3306 add framestepping to playbin2 and seek
3308 2009-06-01 09:59:22 +0200 Peter Kjellerstedt <pkj@axis.com>
3310 * gst-libs/gst/rtsp/gstrtspconnection.c:
3311 rtsp: Avoid compiler warnings with -Wextra.
3313 2009-06-01 09:58:27 +0200 Peter Kjellerstedt <pkj@axis.com>
3315 * gst-libs/gst/rtsp/gstrtspconnection.h:
3316 rtsp: Include gst/gstconfig.h to make sure GST_PADDING is defined.
3318 2009-06-01 09:43:04 +0200 Peter Kjellerstedt <pkj@axis.com>
3320 * gst-libs/gst/sdp/gstsdpmessage.c:
3321 sdp: Remove an unused variable.
3323 2009-05-30 14:17:41 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3325 * gst/ffmpegcolorspace/imgconvert.c:
3326 * gst/ffmpegcolorspace/imgconvert_template.h:
3327 ffmpegcolorspace: Add a lot more conversions from/to 16 bit grayscale
3329 2009-05-29 00:09:15 +0100 Jan Schmidt <thaytan@noraisin.net>
3331 * gst/playback/gstplaybin2.c:
3332 playbin2: Have playbin recognise PGS subpicture streams
3333 Recognise PGS subpicture streams and connect them to the SPU pad
3334 in playsink. Unfortunately this fails badly with negotiation errors
3335 if the SPU is not recent enough to support the stream. I'm not sure
3336 how to add format negotiation in yet.
3338 2009-05-21 23:11:29 +0100 Jan Schmidt <thaytan@noraisin.net>
3340 * gst/playback/gstdecodebin2.c:
3341 * gst/playback/gsturidecodebin.c:
3342 decodebin/uridecodebin: Recognise subpicture/x-pgs pads and output them.
3344 2009-05-28 20:37:59 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
3346 * gst/playback/gstplaysink.c:
3347 playbin2: fix volume handling for audio sinks without "volume" property
3348 When using an audio sink without a "volume" property, volume control
3349 would only work for the first song. For the next song, we'd try to
3350 re-use the existing audio chain, but inadvertently set chain->volume
3351 to NULL instead of to the existing volume element.
3353 2009-05-28 17:05:55 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
3355 * gst/playback/gstplaysink.c:
3356 playbin2: cosmetic change to avoid unnecessary line breaks
3357 Looks nicer and works around gst-indent silliness.
3359 2009-05-28 17:21:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3361 * gst/playback/gstplaysink.c:
3362 playbin2: don't lose the ref to the volume element
3363 Only release the ref to the volume element when it is controled by a sink. For
3364 software volume we never have to fear that it will change.
3366 2009-05-28 15:21:42 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
3368 * gst/playback/gstplaybin2.c:
3369 * gst/playback/gstplaysink.c:
3370 playbin2: actually use configured audio/video sinks
3371 playbin2 inadvertently used autoaudiosink and autovideosink up to now,
3372 since it would overwrite the sinks configured via the "audio-sink"
3373 and "video-sink" properties with the stream-specific group sinks when
3374 configuring the outputs. Those are usually NULL however, so that would
3375 overwrite the configured sinks with NULL which makes playbin2 then
3376 default to the auto sinks. Fix this by keeping a reference to each
3377 configured sink in playbin2 and setting up the right sinks depending
3378 on whether there is a stream-specific sink or not.
3381 2009-05-27 17:37:38 +0300 Stefan Kost <ensonic@users.sf.net>
3383 * tests/examples/seek/seek.c:
3384 seek: add volume label and sync with sink volume
3385 Look at the volume and have the pulsemixer open at same time. Unfortunately
3386 playbin2 does not emit notify on volume right, so this polls for now.
3388 2009-05-27 18:12:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3390 * gst/playback/gstdecodebin2.c:
3391 decodebin2: remove leftover elements
3392 Remove all of the elements inside decodebin2 when goint to READY and NULL.
3393 Makes decodebin2 reusable.
3396 2009-05-27 15:36:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3398 * gst/playback/gstplaysink.c:
3399 playbin2; release refs to volume/mute properties
3400 Release the refs to the volume and mute property elemens before setting the
3401 child elements to READY or NULL.
3404 2009-05-27 12:10:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3406 * gst/gdp/gstgdppay.c:
3407 gdppay: set caps on outgoing buffers
3408 Set caps on outgoing buffers because NULL caps confuse basetransform.
3411 2009-05-27 11:08:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3413 * gst-libs/gst/netbuffer/gstnetbuffer.c:
3414 netbuffer: also note the order of IP4 addresses
3415 IP4 addresses are also stored in network byte order. Make a note of this in the
3418 2009-05-26 22:43:34 +0200 Alessandro Decina <alessandro.d@gmail.com>
3420 * ext/theora/theoraparse.c:
3421 theoraparse: fix assertions in make_granulepos when using the new theora granulepos mapping. Fixes #583903.
3423 2009-05-26 11:13:35 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
3425 * gst-libs/gst/rtsp/gstrtspconnection.c:
3426 Revert "rtspconnection: don't use GLib-2.16 API, we require only 2.14"
3427 This reverts commit 418760cf740332c12c3fd9cf3244af134fa9534b.
3428 We now require GLib 2.16.
3430 2009-05-26 15:18:09 +0100 Jan Schmidt <thaytan@noraisin.net>
3435 2009-05-26 15:37:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3437 * gst-libs/gst/netbuffer/gstnetbuffer.c:
3438 netbuffer: document that the port is network order
3439 Document the fact that we store the port number in network order in
3440 GstNetAddress and that the caller should byteswap appropriately.
3442 2009-05-26 15:23:45 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3444 * gst/videoscale/gstvideoscale.c:
3445 * gst/videoscale/vs_4tap.c:
3446 * gst/videoscale/vs_4tap.h:
3447 * gst/videoscale/vs_image.c:
3448 * gst/videoscale/vs_image.h:
3449 * gst/videoscale/vs_scanline.c:
3450 * gst/videoscale/vs_scanline.h:
3451 videoscale: Add support for 16 bit grayscale in native endianness
3453 2009-05-26 14:58:28 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3455 * gst/ffmpegcolorspace/avcodec.h:
3456 * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
3457 * gst/ffmpegcolorspace/imgconvert.c:
3458 ffmpegcolorspace: Add support for 16 bit grayscale in little/big endian
3460 2009-05-26 14:38:43 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3462 * gst/videotestsrc/videotestsrc.c:
3463 * gst/videotestsrc/videotestsrc.h:
3464 videotestsrc: Add support for 16 bit grayscale in native endianness
3466 2009-01-21 12:33:59 +0100 Andy Wingo <wingo@oblong.net>
3468 add can-activate-pull property to baseaudiosink
3469 * gst-libs/gst/audio/gstbaseaudiosink.c: Add can-activate-pull property
3472 2009-05-26 13:14:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3474 * ext/ogg/gstoggdemux.c:
3475 oggdemux: fix boundary case for seeking.
3476 When we have exactly 0 bytes left to search, make sure we stop instead of going
3477 into an infinite loop.
3479 2009-05-26 11:11:03 +0200 Bastien Nocera <hadess at hadess.net>
3481 * gst-libs/gst/cdda/Makefile.am:
3482 * gst-libs/gst/cdda/gstcddabasesrc.c:
3483 * gst-libs/gst/cdda/sha1.c:
3484 * gst-libs/gst/cdda/sha1.h:
3485 cddabasesrc: Remove copy of sha1 digest
3486 Remove our copy of sha1 digest now that we depend on glib 2.16.
3489 2009-05-25 17:54:01 +0100 Christian Schaller <christian.schaller@collabora.co.uk>
3491 * gst-plugins-base.spec.in:
3494 2009-05-23 00:33:04 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
3496 * gst-libs/gst/video/gstbasevideodecoder.c:
3497 * gst-libs/gst/video/gstbasevideoparse.c:
3498 * gst-libs/gst/video/gstbasevideoutils.c:
3499 * gst-libs/gst/video/gstbasevideoutils.h:
3500 * win32/common/libgstvideo.def:
3501 video: don't expose internal gst_adapter_get_buffer() helper function
3502 If it's really needed it should go into GstAdapter in core.
3504 2009-05-22 21:29:51 -0700 David Schleef <ds@schleef.org>
3506 * gst-libs/gst/video/gstbasevideodecoder.c:
3507 basevideo: Fix memleak
3509 2009-05-22 21:27:58 -0700 David Schleef <ds@schleef.org>
3511 * ext/schroedinger/gstschrodec.c:
3512 * ext/schroedinger/gstschroparse.c:
3513 schro: Fix usage of adapter_masked_scan_uint32
3514 Because *somebody* changed the API without telling me.
3516 2009-05-22 21:25:06 -0700 David Schleef <ds@schleef.org>
3518 * ext/schroedinger/gstschro.c:
3519 schro: Change package name to GST_PACKAGE_NAME
3521 2009-05-22 17:34:10 -0700 David Schleef <ds@schleef.org>
3523 * gst-libs/gst/video/gstbasevideoencoder.c:
3524 basevideo: Add preset interface to encoder
3526 2009-05-22 17:31:14 -0700 David Schleef <ds@schleef.org>
3528 * gst/audioresample/gstaudioresample.c:
3529 Run liboil benchmark multiple times
3530 The statistics function requires multiple runs, otherwise
3531 it causes a divide by zero error.
3533 2009-05-22 19:36:06 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
3535 * m4/gst-fionread.m4:
3536 m4: fix 'suspicious cache value' warning for gst-fionread.m4
3537 .. here as well (should really be moved to common, but I'm too lazy).
3539 2009-05-22 17:41:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3541 * ext/vorbis/vorbisdec.c:
3542 vorbisdec: detect and report errors better
3543 Check the return values of a couple more libvorbis functions and post an error
3544 when something is wrong instead of continuing and crashing.
3546 2009-05-22 15:49:14 +0300 Stefan Kost <ensonic@users.sf.net>
3548 * gst/playback/gstplaysink.c:
3549 playbin2: fix initial volume and mute handling
3550 Use two flags to remember volume/mute changes at times when we don't have the
3551 audiochain yet (e.g. construction). Only set values when they were actualy
3552 changed. This makes pulseaudio's stream restore functional.
3554 2009-05-22 10:19:51 +0100 Jan Schmidt <thaytan@noraisin.net>
3557 Automatic update of common submodule
3558 From d3a8fab to 888e0a2
3560 2009-05-22 09:03:22 +0100 Jan Schmidt <thaytan@noraisin.net>
3562 * win32/common/libgstvideo.def:
3563 win32: Remove gst_adapter_masked_scan_uint32 from the exports
3565 2009-05-21 10:48:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3567 * gst-libs/gst/audio/gstbaseaudiosink.c:
3568 audiosink: improve debug message
3570 2009-05-19 18:10:55 -0700 Michael Smith <msmith@songbirdnest.com>
3572 * gst-libs/gst/tag/gstid3tag.c:
3573 gstid3tag: Don't extract a track number unless present.
3574 In ID3v1, a track number is present only if byte 125 is null AND
3575 byte 126 is non-null. If the track number is not present, don't add
3576 a track number tag with value 0.
3578 2009-05-20 00:48:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3580 * gst-libs/gst/video/gstbasevideoutils.c:
3581 * gst-libs/gst/video/gstbasevideoutils.h:
3582 videoutils: remove adapter methods
3583 Remove adapter methods now that they are in core.
3585 2009-05-20 00:42:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3587 * win32/common/libgstvideo.def:
3588 defs: add new symbols
3590 2009-05-19 17:47:34 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
3593 autogen: pass -Wno-portability to automake to suppress warnings
3596 2009-05-19 02:28:20 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
3598 * docs/libs/.gitignore:
3599 gitignore: remove bogus *.sgml wildcard - these files are tracked in git
3601 2009-05-19 18:41:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3603 * gst/tcp/gsttcpclientsrc.c:
3604 tcpclientsrc: this is not a live source
3605 Don't mark us as a live source because we are not.
3607 2009-05-19 18:41:02 +0300 Stefan Kost <ensonic@users.sf.net>
3609 * gst/adder/gstadder.c:
3610 adder: only send flush_stop when seek failed
3611 This is still not the ultimate fix. Added some comment to explain the troubles.
3613 2009-05-19 17:17:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3615 * gst-libs/gst/audio/gstbaseaudiosink.c:
3616 audiosink: return the return value of wait_preroll
3617 Return the value that _wait_preroll() returned instead of always WRONG_STATE.
3619 2009-05-19 16:45:56 +0300 Stefan Kost <ensonic@users.sf.net>
3621 * gst/adder/gstadder.c:
3622 * gst/adder/gstadder.h:
3623 adder: send flush_stop to match flush_start
3624 Adder was relying that something else sends a flush stop. When using adder with
3625 a livesource it was not getting a flush_stop and thus all pads downstream where
3626 keept flushing. Mark a pending flush_stop and send it when we are working on
3627 the new segment back in the streaming thread.
3629 2009-05-19 16:02:44 +0300 Stefan Kost <ensonic@users.sf.net>
3631 * tests/examples/seek/seek.c:
3632 seek: ui improvements
3633 Repaint the window black on expose, as this looks nicer when resizing or using
3634 the expander. Also show time after slider, as this saves a whole line (nice on
3637 2009-04-29 18:36:17 +0300 Stefan Kost <ensonic@users.sf.net>
3639 * gst/playback/gstdecodebin.c:
3640 decodebin: use iterators instead of list
3641 The list api is deprecated. Use threadsafe iterators instead.
3643 2009-05-19 15:35:54 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3645 * gst/playback/gsturidecodebin.c:
3646 uridecodebin: configure caps on decodebin2
3647 Implement the caps property by setting the configured caps on new decodebin2
3651 2009-05-19 15:34:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3653 * gst/playback/gstdecodebin2.c:
3654 decodebin2: avoid some _caps_ref in some cases
3655 Only mess with the caps refcount when we configure different caps.
3657 2009-05-19 15:27:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3659 * gst/playback/gsturidecodebin.c:
3660 uridecodebin: fix potential caps leak
3661 Free the user-configured caps in finalize.
3663 2009-05-19 15:20:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3665 * gst/playback/gsturidecodebin.c:
3666 uridecodebin: add queue after cdda://
3667 Add a queue2 after the raw output pads of certain sources such as those for uris
3669 No tuning of the queue is done yet as the defaults seem to work fine for me.
3672 2009-05-19 12:45:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3674 * ext/ogg/gstoggdemux.c:
3675 oggdemux: don't loop when at EOS
3676 When we try to read the last page, don't try to read past the upper boundary, as
3677 this might cause endless loops.
3680 2009-05-19 11:20:19 +0200 Edward Hervey <bilboed@bilboed.com>
3682 * gst/audioresample/gstaudioresample.c:
3683 audioresample: Don't drain remaining buffers after a flush.
3684 If we were resetted (due to a flush), we can not drain the remaining
3685 buffers since they would be pushed before a valid new newsegment event.
3687 2009-05-18 22:29:07 -0700 Michael Smith <msmith@syncword.(none)>
3689 * ext/theora/theoradec.c:
3690 theoradec: for 4:2:2, use Y42B (planar) rather than a packed format.
3692 2009-05-19 01:13:34 +0300 Stefan Kost <ensonic@users.sf.net>
3694 * gst/adder/gstadder.c:
3695 adder: add more logging and return value checking
3697 2009-05-19 01:11:45 +0300 Stefan Kost <ensonic@users.sf.net>
3699 * gst/adder/gstadder.c:
3700 adder: handle the return value from iterator_fold
3702 2009-05-19 01:03:44 +0300 Stefan Kost <ensonic@users.sf.net>
3704 * gst/adder/gstadder.c:
3705 adder: use the pad in logging as objects
3706 Helps to differenciate between source and sinks pads.
3708 2009-04-21 22:54:19 +0300 Stefan Kost <ensonic@users.sf.net>
3710 * tests/examples/seek/seek.c:
3711 seek: use parser for mp3 and rename variable
3713 2009-05-18 11:08:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3715 * tests/examples/seek/seek.c:
3716 seek: add playbin2 options in expander
3717 Add the playbin2 stream selection options inside an expander to preserve some
3720 2009-02-10 15:29:10 -0800 David Schleef <ds@schleef.org>
3722 * gst/videotestsrc/videotestsrc.c:
3723 videotestsrc: Add support for v210 and v216 formats
3725 2009-05-15 16:21:15 -0700 David Schleef <ds@schleef.org>
3727 * gst-libs/gst/video/gstbasevideocodec.c:
3728 * gst-libs/gst/video/gstbasevideodecoder.c:
3729 * gst-libs/gst/video/gstbasevideoencoder.c:
3730 * gst-libs/gst/video/gstbasevideoparse.c:
3731 video: remove // comments
3733 2009-05-15 16:18:18 -0700 David Schleef <ds@schleef.org>
3735 * gst-libs/gst/video/video.c:
3736 * gst-libs/gst/video/video.h:
3737 video: Add Y444, v210, v216 formats
3739 2009-05-15 16:12:37 -0700 David Schleef <ds@schleef.org>
3743 * ext/schroedinger/Makefile.am:
3744 * ext/schroedinger/gstschro.c:
3745 * ext/schroedinger/gstschrodec.c:
3746 * ext/schroedinger/gstschroenc.c:
3747 * ext/schroedinger/gstschroparse.c:
3748 * ext/schroedinger/gstschroutils.c:
3749 * ext/schroedinger/gstschroutils.h:
3750 schro: Move schro plugin from Schroedinger
3751 Previous history is in Schroedinger. Depends on, and is an example
3752 of using, GstBaseVideo* base classes.
3753 Code was reindented, and an #ifdef HAVE_ENCODER removed.
3755 2009-05-15 10:23:08 -0700 David Schleef <ds@schleef.org>
3757 * gst-libs/gst/video/Makefile.am:
3758 * gst-libs/gst/video/gstbasevideocodec.c:
3759 * gst-libs/gst/video/gstbasevideocodec.h:
3760 * gst-libs/gst/video/gstbasevideodecoder.c:
3761 * gst-libs/gst/video/gstbasevideodecoder.h:
3762 * gst-libs/gst/video/gstbasevideoencoder.c:
3763 * gst-libs/gst/video/gstbasevideoencoder.h:
3764 * gst-libs/gst/video/gstbasevideoparse.c:
3765 * gst-libs/gst/video/gstbasevideoparse.h:
3766 * gst-libs/gst/video/gstbasevideoutils.c:
3767 * gst-libs/gst/video/gstbasevideoutils.h:
3768 video: Copy BaseVideo classes from Schroedinger
3770 2009-05-15 23:05:45 +0200 Arnout Vandecappelle <arnout@mind.be>
3772 * gst/tcp/gstmultifdsink.c:
3773 multifdsink: add num-fds property
3774 multifdsink::num-fds
3776 2009-05-15 20:36:29 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
3778 * gst-libs/gst/pbutils/descriptions.c:
3779 pbutils: add descriptions for 3GP, JPEG 2000 and Motion JPEG 2000
3781 2009-05-14 11:44:27 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3783 * ext/vorbis/vorbisenc.c:
3784 vorbisenc: Implement Preset interface
3786 2009-05-14 11:43:07 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3788 * ext/theora/theoraenc.c:
3789 theoraenc: Implement Preset interface
3791 2009-05-14 11:41:13 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3793 * ext/ogg/gstoggmux.c:
3794 oggmux: Implement Preset interface
3796 2009-05-14 21:37:22 +0100 Jan Schmidt <thaytan@noraisin.net>
3798 * gst/playback/gstplaysink.c:
3799 playbin2: Fix cdda:// playback
3800 Don't send async-start when the playsink has already been configured
3801 before changing state.
3803 2009-05-14 01:31:48 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
3806 configure: require core CVS for gst_adapter_prev_timestamp()
3807 which is used in the libvisual plugin.
3809 2009-04-22 18:34:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3812 AUTHORS: fix my email
3814 2009-04-22 18:35:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3816 * gst-libs/gst/audio/gstaudioclock.c:
3817 audioclock: make our internal time monotonic
3818 Make the internal time increase monotonically.
3820 2009-05-13 19:27:54 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3822 * ext/libvisual/visual.c:
3823 visual: remove next_ts variable
3824 We can remove the next_ts variable as we don't use it anymore.
3826 2009-05-13 19:24:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3828 * ext/libvisual/visual.c:
3829 visual: use new adapter timestamp code
3830 Use the new adapter timestamp tracking code to make things easier and produce
3831 vastly better output timestamps.
3833 2009-05-13 01:35:07 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
3865 po: avoid conflicts of local *.po files with files in git
3866 Make it so that filenames and line numbers are only stored in the *.pot file
3867 (which is not in git), but not in the individual *.po files. This information
3868 is hardly useful for translators in our case, and it should avoid the constant
3869 conflicts of local *.po files with the ones in git which are caused by the
3870 source files changing and the line numbers being updated. This commit might
3871 cause one last merge conflict for you, which you can work around with
3872 "git checkout po/*.po" before merging or pulling. After that there should
3873 (hopefully) not be any more local modifications of these files (unless
3874 someone committed additions or changes to translated strings and the
3875 *.po files haven't been updated yet, that is).
3877 2009-05-12 23:51:08 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
3879 * tests/check/elements/.gitignore:
3880 * tests/check/elements/audioresample.c:
3881 tests: fix audioresample unit test on big endian architectures
3882 Don't hardcode endianness=1234 in the filtercaps, it will cause
3883 pad link failures which will result in the test timing out.
3885 2009-05-12 17:18:37 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
3887 * gst/audiotestsrc/gstaudiotestsrc.c:
3888 audiotestsrc: fix broken enum nick - it should have a hyphen
3889 The enum nick should be 'sine-table', not 'sine table'. Technically this is
3890 an API/ABI change I guess, but anyone who was using this and didn't report
3893 2009-05-01 01:04:48 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
3895 * gst/audiotestsrc/gstaudiotestsrc.c:
3896 audiotestsrc: seek to the requested byte offset, not the expected byte offset
3898 2009-05-01 01:03:06 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
3900 * gst/audiotestsrc/gstaudiotestsrc.c:
3901 * gst/audiotestsrc/gstaudiotestsrc.h:
3902 audiotestsrc: support more than just one channel
3904 2009-05-12 15:52:41 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3906 * gst-libs/gst/interfaces/propertyprobe.h:
3907 propertyprobe: Fix typo in the docs
3909 2009-05-12 12:17:55 +0100 Christian Schaller <christian.schaller@collabora.co.uk>
3911 * ext/ogg/gstoggmux.c:
3912 * ext/theora/theora.c:
3913 * ext/vorbis/vorbis.c:
3914 Add ranks to the Oggmuxer, Vorbis encoder and Theora encoder
3916 2009-04-30 16:37:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3918 * gst/videorate/gstvideorate.c:
3919 * gst/videorate/gstvideorate.h:
3920 videorate: handle invalid timestamps better
3921 Handle buffers with -1 timestamps better by keeping track of the en time of the
3922 previous buffer and assuming the -1 timestamp buffer goes right after the
3924 when we have two buffers that are equally good, output the oldest buffer once to
3926 don't try to calculate latency when the input framerate is unknown.
3928 2009-04-28 11:37:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3930 * ext/ogg/gstoggmux.c:
3931 oggmux: small debug statement in DISCONT
3933 2009-04-28 11:24:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3935 * ext/ogg/gstoggdemux.c:
3936 * ext/ogg/gstoggdemux.h:
3937 oggdemux: fix abuse of ogg API, handle broken oggs
3938 When we feed the ogg sync layer, we need to feed it contiguous data even if the
3939 sync layer did not consume all of it yet. This makes sure that it always finds
3940 the next page even for more corrupted files. Use a different read_offset for
3941 this purpose. since we now keep track of the sync layer, we don't have to reset
3942 after finding a start of a page.
3943 Add some more debug info for the error paths.
3944 Only reset the sync layer when we perform a seek operation.
3945 Avoid failure when the next chain has no bos pages but instead simply ignore it.
3946 when we receive unknown page serial numbers mid stream, don't fail but post a
3947 warning and hope that we get back on track later.
3950 2009-04-30 16:41:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3952 * gst/playback/gstdecodebin2.c:
3953 decodebin2: make subpictures a raw output format
3954 Subpictures are a raw format, we want those pads exposed so that playbin2 can do
3955 the subpicture mixing.
3957 2009-04-27 10:15:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3959 * gst-libs/gst/rtp/gstbasertppayload.c:
3960 * gst-libs/gst/rtp/gstbasertppayload.h:
3961 rtpdepay: add some more comments
3963 2009-04-17 10:54:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3965 * gst-libs/gst/audio/gstaudioclock.c:
3966 audioclock: make sure values are ever increasing
3968 2009-05-05 17:17:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3970 * gst/playback/gstplaysink.c:
3971 playbin2: make fallback identity silent
3972 Set the signal-handoffs to FALSE and silent to TRUE for the fallback identity
3973 element so that it consumes less CPU.
3975 2009-04-17 10:57:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3977 * gst/playback/gstplaybin2.c:
3978 * gst/playback/gstplaysink.c:
3979 playbin2: handle custom audiosinks differently
3980 Keep track of the autoplugged custom sinks and configure them in the playsink
3981 element when we have collected all streams.
3982 Also make sure that we only select one custom sink.
3983 When unreffing the internal sink, we don't need to change the state to NULL.
3985 2009-05-12 10:36:25 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
3987 * gst/playback/gstplaybin2.c:
3988 * gst/playback/gstplaysink.c:
3989 * gst/playback/gstplaysink.h:
3990 playbin2: unify custom sink get/set functions
3991 Use one function to set/get all of the different sink types.
3992 cleanup up the subpicture chain too.
3993 Allow setting a custom subpicture sink.
3995 2009-05-11 18:29:34 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
3997 * gst-libs/gst/interfaces/tunernorm.h:
3998 interfaces: Seperate some more struct definitions from typedefs
4000 2009-05-11 15:48:56 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4002 * gst-libs/gst/interfaces/navigation.h:
4003 * gst-libs/gst/interfaces/videoorientation.h:
4004 * gst-libs/gst/interfaces/xoverlay.h:
4005 interfaces: Seperate some more struct definitions from typedefs
4007 2009-05-10 17:28:53 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4009 * win32/common/libgstinterfaces.def:
4010 Add new functions to win32 exports
4012 2009-05-10 17:28:05 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4014 * docs/libs/gst-plugins-base-libs-sections.txt:
4015 Add new functions to the docs
4017 2009-05-10 17:25:58 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4019 * gst-libs/gst/interfaces/mixer.c:
4020 * gst-libs/gst/interfaces/mixer.h:
4021 interfaces: API: Add gst_mixer_get_mixer_type()
4022 This is a convenience function that returns the mixer_type
4023 of the interface struct.
4025 2009-05-10 17:25:31 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4027 * gst-libs/gst/interfaces/colorbalance.c:
4028 interfaces: Add docs for gst_color_balance_get_balance_type()
4030 2009-05-10 11:17:19 +0200 Marc-Andre Lureau <marcandre.lureau@gmail.com>
4033 Run libtoolize before aclocal
4034 This unbreaks the build in some cases. Fixes bug #582021
4036 2009-05-07 17:38:40 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4038 * ext/pango/gsttextrender.c:
4039 textrender: Correctly initialize the background for ARGB too
4041 2009-05-07 16:59:32 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4043 * ext/pango/gsttextrender.c:
4044 * ext/pango/gsttextrender.h:
4045 textrender: Use libgstvideo functions to create caps
4046 Also check if downstream wants ARGB always when we get
4049 2009-05-07 16:52:02 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4051 * ext/pango/gsttextrender.c:
4052 textrender: Don't always use ARGB if downstream supports it but take it's preference
4054 2009-05-07 16:48:08 +0200 Kapil Agrawal <kapil@mediamagictechnologies.com>
4056 * ext/pango/gsttextrender.c:
4057 * ext/pango/gsttextrender.h:
4058 textrender: Add support for ARGB and alignment properties
4061 2009-05-07 16:42:20 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4063 * ext/pango/gsttextrender.c:
4064 textrender: Add ; after GST_BOILERPLATE to fix indention
4066 2009-05-07 15:10:30 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4068 * gst-libs/gst/tag/gstvorbistag.c:
4069 vorbistag: Use text/uri-list as mimetype instead of ---> for URI lists
4071 2009-05-07 14:59:36 +0200 Arnout Vandecappelle <arnout@mind.be>
4073 * gst/typefind/gsttypefindfunctions.c:
4074 typefindfunctions: made mp3_type_find less aggressive
4075 mp3_type_find could suggest already when only a single valid header
4076 was found, if it ran out of data before the end of the next frame.
4077 Therefore, ignore the last found frame if it was incomplete.
4080 2009-05-07 14:48:29 +0200 John Millikin <jmillikin@gmail.com>
4082 * gst-libs/gst/tag/gstvorbistag.c:
4083 vorbistag: Store cover art in vorbiscomments
4086 2009-05-07 06:14:18 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4088 * gst-libs/gst/interfaces/colorbalance.c:
4089 * gst-libs/gst/interfaces/colorbalance.h:
4090 interfaces: API: Add gst_color_balance_get_balance_type()
4091 This is a convenience function that returns the balance_type
4092 of the interface struct.
4094 2009-05-06 17:59:13 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4096 * gst-libs/gst/interfaces/colorbalance.h:
4097 * gst-libs/gst/interfaces/colorbalancechannel.h:
4098 * gst-libs/gst/interfaces/tuner.h:
4099 * gst-libs/gst/interfaces/tunerchannel.h:
4100 interfaces: Separate struct definitions from typedefs
4102 2009-05-06 14:03:01 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4104 * pkgconfig/gstreamer-app-uninstalled.pc.in:
4105 Fix libdir for uninstalled gstreamer-app library
4107 2009-05-12 01:59:01 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
4109 * gst-libs/gst/pbutils/descriptions.c:
4110 pbutils: add description for APE tag caps
4112 2009-05-12 01:35:27 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
4115 configure: bump core requirement to last release
4116 as that's more likely to be true than that we need
4119 2009-05-12 01:21:57 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
4123 configure: rename CVS -> git in a couple of places
4125 2009-05-12 01:17:53 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
4128 configure: bump GLib requirement to GLib >= 2.16
4129 as per the New Regime (see wiki).
4131 2009-05-01 00:09:15 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
4133 * gst-libs/gst/tag/gsttagdemux.c:
4134 tagdemux: cache events from upstream and re-send them once we have a source pad
4135 Makes sure tags don't get dropped when we have multiple tag demuxers in a row.
4138 2009-05-07 14:07:44 -0700 Michael Smith <msmith@songbirdnest.com>
4140 * gst-libs/gst/riff/riff-media.c:
4141 riff: support UYVY raw 4:2:2 in riff.
4143 2009-05-11 21:20:07 +0100 Jan Schmidt <thaytan@noraisin.net>
4146 Back to development -> 0.10.23.1
4148 2009-04-27 22:42:55 -0700 Michael Smith <msmith@syncword.(none)>
4150 * ext/theora/theoradec.c:
4151 theoradec: fix buffer overrun on 422 decode.
4153 2009-04-27 21:39:01 -0700 Michael Smith <msmith@syncword.(none)>
4155 * ext/theora/theoradec.c:
4156 theoradec: 444 support.
4158 2009-04-27 21:30:04 -0700 Michael Smith <msmith@syncword.(none)>
4160 * ext/theora/theoradec.c:
4161 theoradec: handle 422 images (as YUY2).
4163 2009-04-27 21:01:51 -0700 Michael Smith <msmith@syncword.(none)>
4165 * ext/theora/gsttheoradec.h:
4166 * ext/theora/theoradec.c:
4167 theoradec: rearrange code in preparation for 422 and 444 support.
4169 === release 0.10.23 ===
4171 2009-05-10 23:57:01 +0100 Jan Schmidt <thaytan@noraisin.net>
4177 * docs/plugins/gst-plugins-base-plugins.args:
4178 * docs/plugins/gst-plugins-base-plugins.hierarchy:
4179 * docs/plugins/gst-plugins-base-plugins.interfaces:
4180 * docs/plugins/gst-plugins-base-plugins.prerequisites:
4181 * docs/plugins/gst-plugins-base-plugins.signals:
4182 * docs/plugins/inspect/plugin-adder.xml:
4183 * docs/plugins/inspect/plugin-alsa.xml:
4184 * docs/plugins/inspect/plugin-app.xml:
4185 * docs/plugins/inspect/plugin-audioconvert.xml:
4186 * docs/plugins/inspect/plugin-audiorate.xml:
4187 * docs/plugins/inspect/plugin-audioresample.xml:
4188 * docs/plugins/inspect/plugin-audiotestsrc.xml:
4189 * docs/plugins/inspect/plugin-cdparanoia.xml:
4190 * docs/plugins/inspect/plugin-decodebin.xml:
4191 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
4192 * docs/plugins/inspect/plugin-gdp.xml:
4193 * docs/plugins/inspect/plugin-gio.xml:
4194 * docs/plugins/inspect/plugin-gnomevfs.xml:
4195 * docs/plugins/inspect/plugin-libvisual.xml:
4196 * docs/plugins/inspect/plugin-ogg.xml:
4197 * docs/plugins/inspect/plugin-pango.xml:
4198 * docs/plugins/inspect/plugin-playback.xml:
4199 * docs/plugins/inspect/plugin-queue2.xml:
4200 * docs/plugins/inspect/plugin-subparse.xml:
4201 * docs/plugins/inspect/plugin-tcp.xml:
4202 * docs/plugins/inspect/plugin-theora.xml:
4203 * docs/plugins/inspect/plugin-typefindfunctions.xml:
4204 * docs/plugins/inspect/plugin-uridecodebin.xml:
4205 * docs/plugins/inspect/plugin-video4linux.xml:
4206 * docs/plugins/inspect/plugin-videorate.xml:
4207 * docs/plugins/inspect/plugin-videoscale.xml:
4208 * docs/plugins/inspect/plugin-videotestsrc.xml:
4209 * docs/plugins/inspect/plugin-volume.xml:
4210 * docs/plugins/inspect/plugin-vorbis.xml:
4211 * docs/plugins/inspect/plugin-ximagesink.xml:
4212 * docs/plugins/inspect/plugin-xvimagesink.xml:
4213 * gst-plugins-base.doap:
4214 * win32/common/_stdint.h:
4215 * win32/common/config.h:
4218 2009-05-10 23:56:05 +0100 Jan Schmidt <thaytan@noraisin.net>
4251 2009-05-08 20:32:20 +0100 Jan Schmidt <thaytan@noraisin.net>
4283 * win32/common/_stdint.h:
4284 * win32/common/config.h:
4285 0.10.22.6 pre-release
4287 2009-05-08 13:09:32 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4289 * gst/playback/gstplaysink.c:
4290 playbin2: fix resume after pause
4291 Don't ignore the state change of the children, they might be doing an ASYNC
4294 2009-05-08 11:05:41 +0100 Jan Schmidt <thaytan@noraisin.net>
4327 0.10.22.5 pre-release
4329 2009-05-07 22:01:01 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4331 * gst/tcp/gstmultifdsink.c:
4332 * gst/tcp/gsttcp-marshal.list:
4333 multifdsink: fix signature of the add-full signal
4334 The second parameter is a GstSyncMethod enum, not a boolean.
4336 2009-05-07 15:19:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4338 * gst/playback/gstplaysink.c:
4339 playsink: initialize variable too
4341 2009-05-07 14:28:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4343 * gst/playback/gstplaysink.c:
4344 playbin2: make playsink go ASYNC to PAUSED
4345 Make playsink go async to the PAUSED state instead of relying on uridecodebin
4346 for async behaviour in playbin. This solves some problems (mainly with DVD)
4347 where the pipeline would go to PLAYING before preroll completed, failing to
4348 select the audiosink clock.
4351 2009-05-06 16:09:52 +0100 Jan Schmidt <thaytan@noraisin.net>
4383 * win32/common/_stdint.h:
4384 * win32/common/config.h:
4385 0.10.22.4 pre-release
4387 2009-05-06 13:19:34 +0100 Zaheer Merali <zaheerabbas@merali.org>
4389 * ext/theora/theoraenc.c:
4390 * ext/vorbis/vorbisenc.c:
4391 vorbisenc, theoraenc: Ensure gp is computed consistently + clip to segment
4392 With vorbisenc, compute the granulepos with running time and clip incoming
4394 With theoraenc, drop out of segment buffers.
4396 2009-05-01 16:47:53 +0100 Jan Schmidt <thaytan@noraisin.net>
4398 * gst/audioresample/gstaudioresample.c:
4399 audioresample: Fix buffer size transformations
4400 When calculating the input/output buffer sizes in the transform_size function,
4401 take the number of channels into account, so we don't end up calculating
4402 a buffer size that only contains a partial number of audio frames.
4403 Also, when going from output size to input size, round down rather than
4404 up, so as to calculate the minimum number of samples that *might* yield
4405 a buffer of the intended destination size.
4406 Fixes: #580470 and #580952
4408 2009-04-29 16:45:27 +0100 Jan Schmidt <thaytan@noraisin.net>
4410 * ext/vorbis/gstvorbisenc.h:
4411 * ext/vorbis/vorbisenc.c:
4412 vorbisenc: Ensure output buffers fall within the segment
4413 Add the start position of the first segment to the running time
4414 used to generate buffer timestamps in vorbisenc. This avoids generating
4415 buffers which fall outside the initial segment. The element segment
4416 handling requires more extensive fixing, but this at least prevents
4417 regressions. Fixes: #580020
4419 2009-04-29 11:18:42 +0200 Andy Wingo <wingo@oblong.net>
4421 * gst-libs/gst/audio/gstbaseaudiosink.c:
4422 Revert "add can-activate-pull property to baseaudiosink"
4423 This reverts commit c4074a2ee4f1e6cac734a145bf675bbb16fac985.
4425 2009-04-29 11:18:33 +0200 Andy Wingo <wingo@oblong.net>
4427 * gst-libs/gst/audio/gstbaseaudiosink.c:
4428 Revert "[baseaudiosink] add docs for can-activate-pull"
4429 This reverts commit 416ce16f26b39c76ab35e1ef6a75dc41ec69f75b.
4431 2009-04-28 18:48:33 +0200 Andy Wingo <wingo@oblong.net>
4433 [baseaudiosink] add docs for can-activate-pull
4434 * gst-libs/gst/audio/gstbaseaudiosink.c: Add documentation for
4437 2009-01-21 12:33:59 +0100 Andy Wingo <wingo@oblong.net>
4439 add can-activate-pull property to baseaudiosink
4440 * gst-libs/gst/audio/gstbaseaudiosink.c: Add can-activate-pull property
4443 2009-04-28 11:32:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4445 * gst/videorate/gstvideorate.c:
4446 * gst/videorate/gstvideorate.h:
4447 videorate: clear discont on duplicated buffers
4448 When videorate duplicates a buffer with a DISCONT flag, it copies the discont on
4449 the first pushed buffer but fails to clear it for subsequent buffers. This
4450 causes theoraenc!oggmux and possibly other elements to consider this a discont
4452 Fix videorate to produce discont as the first buffer and after a flushing seek.
4455 2009-04-24 18:13:00 +0100 Jan Schmidt <thaytan@noraisin.net>
4457 * tests/check/Makefile.am:
4458 check: Disable the playbin2 for this release, as it is a bit racy.
4459 Disable the test, as per the discussion in #580120. Needs re-enabling
4460 after the release, when playbin2 is fixed.
4462 2009-04-23 08:41:19 +0200 Edward Hervey <bilboed@bilboed.com>
4464 * gst/playback/gstdecodebin2.c:
4465 decodebin2: Don't reduce max-size-time of exposed groups. Fixes #579912
4466 The 2s limit is way too small for a lot of files (which have an interleave
4467 in time of between 3 and 5s). Instead, leave it to the initial 5s value
4468 and reduce the other limits (allowing us to stay memory-efficient).
4470 2009-04-21 21:06:59 +0100 Jan Schmidt <thaytan@noraisin.net>
4502 * win32/common/_stdint.h:
4503 * win32/common/config.h:
4504 0.10.22.3 pre-release
4506 2009-04-21 20:41:23 +0100 René Stadler <mail@renestadler.de>
4508 * gst/audioresample/gstaudioresample.c:
4509 audioresample: Fix unused variable in compilation with --disable-gst-debug
4512 2009-04-21 22:12:28 +0100 Jan Schmidt <thaytan@noraisin.net>
4515 Automatic update of common submodule
4516 From b3941ea to 6ab11d1
4518 2009-04-21 20:57:34 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4520 * gst/playback/gstplaybasebin.c:
4521 playbin: only use raw_decoding_mode when it's true
4522 First check the pad caps if they are raw before setting the raw_decoding_mode to
4523 TRUE. Fixes playback of transport streams and other streams that require large
4527 2009-04-19 18:15:28 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
4529 * gst-libs/gst/cdda/gstcddabasesrc.c:
4530 * tests/check/libs/cddabasesrc.c:
4531 cddabasesrc: fix posting of discid tags after MERGE_MODE_REPLACE_ALL changes in core
4532 Don't use REPLACE_ALL merge mode when that's not really what we want,
4533 as now that REPLACE_ALL actually does what it's supposed to do in
4534 core, we drop tags we wanted to keep, such as the various disc id
4535 tags. Add unit test for this as well. Fixes #579463.
4537 2009-04-17 10:34:54 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
4539 * gst-libs/gst/rtsp/gstrtspconnection.c:
4540 rtspconnection: don't use GLib-2.16 API, we require only 2.14
4543 2009-04-17 10:55:08 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4545 * gst-libs/gst/audio/gstbaseaudiosink.c:
4546 baseaudiosink: don't unparent the ringbuffer
4547 when going to NULL, don't unparent the ringbuffer because we don't support going
4548 back to 0 very well yet.
4551 2009-04-17 10:53:10 +0200 Olivier Crete <tester at tester.ca>
4553 * gst-libs/gst/rtp/gstrtcpbuffer.c:
4554 RTCP: don't fail when retrieving invalid PT
4555 We can't meaningfully assert on valid packet types so just return the type as it
4556 is. Update the comments to reflect this.
4559 2009-04-16 12:12:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4561 * docs/libs/gst-plugins-base-libs-sections.txt:
4562 * gst-libs/gst/app/gstappsink.h:
4563 * gst-libs/gst/app/gstappsrc.h:
4564 app: add trivial cast macros
4565 Add trivial cast macros for appsrc and appsink. Mark them as being since 0.10.23
4566 and add the macros to the standard macros in the docs.
4569 2009-04-16 12:09:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4571 * pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
4572 pkgconfig: add the app/ directory to Libs
4573 Add the appsrc/appsink directory to the Libs in the uninstalled
4574 pkgconfig file so that one can build against it.
4577 2009-04-15 22:59:31 +0100 Jan Schmidt <thaytan@noraisin.net>
4580 0.10.22.2 pre-release
4582 2009-04-15 22:56:15 +0100 Jan Schmidt <thaytan@noraisin.net>
4585 ChangeLog: regenerate changelog with the gen-changelog script
4587 2009-04-16 00:41:13 +0100 Jan Schmidt <thaytan@noraisin.net>
4618 po: Update po files from TP
4620 2009-04-16 00:40:59 +0100 Jan Schmidt <thaytan@noraisin.net>
4622 * win32/common/_stdint.h:
4623 * win32/common/config.h:
4624 * win32/common/gstrtsp-enumtypes.c:
4625 * win32/common/interfaces-enumtypes.c:
4626 * win32/common/interfaces-enumtypes.h:
4627 * win32/common/video-enumtypes.c:
4628 win32: Update win32 build files
4630 2009-04-16 00:31:55 +0100 Jan Schmidt <thaytan@noraisin.net>
4632 * tests/check/libs/video.c:
4633 check: Add GST_VIDEO_FORMAT_YVYU to the test so it passes.
4635 2009-04-16 00:31:00 +0100 Jan Schmidt <thaytan@noraisin.net>
4637 * tests/check/elements/playbin2.c:
4638 check: Fix the input uri in playbin2 test.
4639 Don't try and use a random file in wim's home directory as a test input
4641 2009-04-15 15:35:59 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4643 * gst-libs/gst/video/video.h:
4644 video: Fix typo in the docs
4646 2009-04-15 14:53:47 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4648 * gst-libs/gst/video/video.c:
4649 * gst-libs/gst/video/video.h:
4650 video: Add support for YVYU YUV colorspace
4652 2009-04-15 00:17:51 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
4654 * docs/libs/gst-plugins-base-libs-docs.sgml:
4655 * gst-libs/gst/fft/gstfft.c:
4656 docs: fix hyperlink and move fft attribution to the right place
4658 2009-04-15 00:02:39 +0300 Stefan Kost <ensonic@users.sf.net>
4660 * gst-libs/gst/audio/gstbaseaudiosink.c:
4661 log: use G_GUINT64_FORMAT instead of llu
4663 2009-04-14 18:31:52 +0200 Josep Torra <n770galaxy at gmail.com>
4665 * gst-libs/gst/rtsp/gstrtspdefs.c:
4666 * gst-libs/gst/rtsp/gstrtspdefs.h:
4667 RTSP: add missing headers for WMS RTSP
4668 Add missing headers related to Windows Media RTSP extension.
4671 2009-04-14 18:16:37 +0200 Olivier Crete <tester at tester.ca>
4673 * docs/design/draft-keyframe-force.txt:
4674 * ext/theora/gsttheoraenc.h:
4675 * ext/theora/theoraenc.c:
4676 theoraenc: implement upstream keyframe force
4677 Implement handling of upstream keyframe forcing.
4678 Update the design documents too.
4681 2009-04-14 17:31:31 +0200 Olivier Crete <tester at tester.ca>
4683 * ext/theora/theoraenc.c:
4684 theoraenc: factor out keyframe forcing
4687 2009-04-14 17:01:51 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
4690 * gst-libs/gst/fft/gstfft.c:
4691 Give credit to Mark Borgerding (kissfft author)
4692 and add myself to AUTHORS as well. Fixes #575638.
4694 2009-04-14 17:04:06 +0200 Jan Urbanski <j.urbanski at students.mimuw.edu.pl>
4696 * gst/tcp/gstmultifdsink.c:
4697 * gst/tcp/gstmultifdsink.h:
4698 multifdsink: add property to resend streamheaders
4699 Adds a new property in multifdsink, resend-streamheader.
4700 If this property is false, the multifdsink will not send the streamheader if
4701 there's already one set for a particular client.
4702 There are some formats in which every stream needs to start with a certain
4703 blob, but you can't inject this blob at leisure. If the producer wants to
4704 change the blob in question and sets in as the streamheader on the outgoing
4705 buffers' caps, new clients of multifdsink will get the new streamheader, but
4706 old clients will break, because they'll see the blob in the middle of the
4708 The property is true by default, so existing code will not see any difference.
4711 2009-04-14 16:53:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4713 * gst/tcp/gstmultifdsink.c:
4714 * gst/tcp/gstmultifdsink.h:
4715 multifdsink: add property to handle client write
4716 Add a property to disable listening to client writes. This property is usefull
4717 when other code will deal with reading from the client socket.
4718 API: GstMultiFdSink::handle-read property
4720 2009-04-14 16:45:20 +0200 Johann Prieur <johann.prieur at gmail.com>
4722 * docs/libs/gst-plugins-base-libs-sections.txt:
4723 * gst-libs/gst/rtp/gstrtcpbuffer.c:
4724 * gst-libs/gst/rtp/gstrtcpbuffer.h:
4725 * win32/common/libgstrtp.def:
4726 RTCP: add beginnings of Feedback messages
4727 Add the beginnings of parsing and constructing Feedback messages.
4730 2009-04-14 13:51:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4732 * gst/playback/gstplaysink.c:
4733 playbin2: clear the target
4734 Clear the target of our ghostpads before we remove the pad from the element.
4735 This to make sure that the internal pad is not left linked to whatever pad we
4736 were ghosted to. This should only be a problem when we leak the ghostpads.
4737 Also release our subpicture pads.
4740 2009-04-14 12:10:30 +0100 Hannes Bistry <hannesb@gmx.net>
4742 * sys/ximage/ximagesink.c:
4743 ximagesink: fix mouse pointer offsets in navigation event if window is smaller than the image
4746 2009-04-14 13:16:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4748 * gst-libs/gst/audio/gstbaseaudiosrc.c:
4749 baseaudiosrc: adjust the internal timestamp
4750 Adjust the internal timestamp before comparing it against the adjusted clock
4754 2009-04-14 13:12:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4756 * gst-libs/gst/audio/gstbaseaudiosink.c:
4757 baseaudiosink: use new clock time methods
4758 Use the unadjusted internal clock times to calculate the internal/external
4759 offset when calibrating the clock.
4760 When going to NULL, unparent and free the ringbuffer, like we do in the source
4764 2009-04-14 13:08:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4766 * gst-libs/gst/audio/gstaudioclock.c:
4767 * gst-libs/gst/audio/gstaudioclock.h:
4768 * win32/common/libgstaudio.def:
4769 audioclock: add methods for the internal offset
4770 Add two methods for getting the unadjusted time of the clock and one for
4771 adjusting an internal time. We will need these methods for correctly handling
4772 the time after a gst_audio_clock_reset().
4773 Add a debug category and some debug lines to the audio clock.
4774 API: gst_audio_clock_get_time()
4775 API: gst_audio_clock_adjust()
4776 API: GST_AUDIO_CLOCK_CAST()
4778 2009-04-14 11:34:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4780 * gst/playback/gstdecodebin2.c:
4781 decodebin2: fix up the debugs and warnings
4782 Use _OBJECT variants because we can. Go over some log statements and put them in
4786 2009-04-12 22:26:33 +0200 Luca Ognibene <luca.ognibene at gmail.com>
4788 * gst/tcp/gstmultifdsink.c:
4789 multifdsink: fix error in sync-method
4790 Multifdsink did not handle sync-method=latest-keyframe correctly when the
4791 soft-limit is set to -1 (unlimited).
4794 2009-04-10 21:49:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4796 * gst-libs/gst/audio/gstbaseaudiosink.c:
4797 baseaudiosink: use the internal clock time
4798 We can't assume that the internal clock time is the same as the function we
4799 installed on our provided clock because somebody might have changed it.
4801 2009-04-10 14:12:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4803 * tests/examples/seek/seek.c:
4804 seek: handle clock-lost messages
4805 When we receive a clock-lost message we need to pause and play to select a new
4808 2009-04-10 13:44:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4810 * tests/check/Makefile.am:
4811 * tests/check/elements/playbin2.c:
4812 check: add a unit test for playbin2
4813 Add unit test for playbin2 and include the refcount test in #577794.
4815 2009-04-10 13:42:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4817 * gst/playback/gstplaysink.c:
4818 playbin2: fix refcounting of visualisations
4821 2009-04-10 13:27:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4823 * gst/playback/gstplaysink.c:
4824 playsink: fix refcounting of custom elements
4825 Sink the custom sinks, let other elements we create be sunken by the bin we add
4829 2009-04-10 12:27:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4831 * tests/check/elements/appsink.c:
4832 check: fix appsink test
4833 Fix the appsink test now that the method signature changed.
4835 2009-04-10 12:26:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4837 * gst/playback/gstplaybin2.c:
4838 playbin2: handle missing input-selector
4839 Gracefully degrade and disable stream selection when input-selector is
4842 2009-04-09 23:46:17 +0200 Martin Samuelsson <martin.samuelsson at axis.com>
4844 * gst-libs/gst/app/gstappsink.c:
4845 * gst-libs/gst/app/gstappsink.h:
4846 appsink: make callbacks return GstFlowReturn
4847 Make the new_buffer and new_preroll callbacks return a GstFlowReturn so that
4848 errors can be reported properly.
4851 2009-04-09 18:04:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4853 * gst-libs/gst/audio/gstringbuffer.c:
4854 * gst-libs/gst/audio/gstringbuffer.h:
4855 ringbuffer: allow for custom commit functions
4856 Allow subclasses to override the commit method.
4858 2009-04-08 18:04:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4860 * gst-libs/gst/audio/gstbaseaudiosink.c:
4861 baseaudiosink: fix a small glitch after pause
4862 After we pause the stream and interrupt the writeout to the ringbuffer, also adjust
4863 the amount of output samples we consumed. We can't do this reliably with the
4864 current API when we are doing trick modes but we can do the right thing for
4867 2009-04-08 16:43:27 +0300 Stefan Kost <ensonic@users.sf.net>
4869 * gst/playback/gstplaysink.c:
4870 playbin2: better error message on sink failure
4871 If we could create the sinks, but the don't work, don't send the missing plugin
4872 message and report that the state-changed failed.
4874 2009-04-07 22:38:29 +0300 Stefan Kost <ensonic@users.sf.net>
4876 * gst-libs/gst/audio/gstaudiofilter.c:
4877 audiofilter: don't leak pad-template
4878 gst_element_class_add_pad_template() does not take ownership.
4880 2009-04-04 21:18:38 +0300 Felipe Contreras <felipe.contreras@gmail.com>
4883 Automatic update of common submodule
4884 From d0ea89e to b3941ea
4886 2009-04-04 16:28:14 +0200 Edward Hervey <bilboed@bilboed.com>
4888 * gst-libs/gst/interfaces/navigation.c:
4889 * sys/v4l/v4lsrc_calls.c:
4890 navigation/v4l: Don't use g_return_val_if_fail for computed/used values.
4892 2009-03-22 09:46:37 +0100 Edward Hervey <bilboed@bilboed.com>
4894 * ext/theora/theoradec.c:
4895 theoradec: return GST_CLOCK_TIME_NONE for negative framecounts.
4896 This fixes most seeking issues when used with gnonlin.
4899 2009-04-04 14:53:42 +0200 Edward Hervey <bilboed@bilboed.com>
4902 Automatic update of common submodule
4903 From f8b3d91 to d0ea89e
4905 2009-04-03 10:51:42 -0700 Michael Smith <msmith@songbirdnest.com>
4907 * gst/playback/gstplaybin2.c:
4908 playbin2: don't leak selector when getting current stream numbers.
4910 2009-04-02 22:28:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4912 * gst-libs/gst/rtsp/gstrtspconnection.c:
4913 rtsp: use fully qualified urls when using a proxy
4914 Use a fully qualified url when specifying the url for tunneled requests through
4918 2009-03-31 00:54:30 +0100 Jan Schmidt <thaytan@noraisin.net>
4920 * docs/libs/gst-plugins-base-libs-sections.txt:
4921 * gst-libs/gst/interfaces/navigation.c:
4922 * gst-libs/gst/interfaces/navigation.h:
4923 * tests/check/Makefile.am:
4924 * tests/check/libs/.gitignore:
4925 * tests/check/libs/navigation.c:
4926 * win32/common/libgstinterfaces.def:
4927 navigation: Extend the navigation interface
4928 Add support for a set of standard commands that can be queried and executed to
4929 support applications like DVD. Add query construction and parsing functions.
4930 Add new messages that can be sent on the bus to provide notifications related
4931 to commands, multiangle changes, and button highlight activity.
4932 Add some helper functions to parse the existing GstNavigation events that
4933 elements might receive.
4934 Document it all and add unit tests.
4936 2009-02-04 17:03:07 +0000 Jan Schmidt <thaytan@noraisin.net>
4938 * gst/playback/gstplaybasebin.c:
4939 * gst/playback/gstplaybasebin.h:
4940 playbin: Add simple 'raw decoding mode'.
4941 Raw decoding mode removes almost all buffering in video and audio queues
4942 when a source providing already decoded video/audio is detected, on the
4943 possibly bogus assumption that such a source should provide sufficient
4944 internal queueing. Fixes playback on some DVDs, and improves it
4947 2009-04-02 09:27:07 +0100 Jan Schmidt <thaytan@noraisin.net>
4949 * tests/check/elements/.gitignore:
4950 ignores: Ignore the videoscale check binary
4952 2009-04-02 12:13:57 +0100 Jan Schmidt <thaytan@noraisin.net>
4954 * win32/common/libgstrtsp.def:
4955 win32: Add gst_rtsp_connection_set_proxy to the win32 exports
4957 2009-04-02 10:42:19 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
4959 * ext/alsa/gstalsamixer.c:
4960 alsamixer: don't forget to release locks in a few places
4963 2009-04-02 11:10:12 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4965 * gst/videoscale/vs_4tap.c:
4966 videoscale: Don't read over line ends when taking the last Cr or Cb
4968 2009-04-02 10:52:06 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
4970 * gst/videoscale/vs_4tap.c:
4971 videoscale: Don't write to few pixels and don't mix Cr and Cb
4974 2009-04-01 15:15:57 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
4976 * gst/audioresample/gstaudioresample.c:
4977 * tests/check/elements/audioresample.c:
4978 audioresample: fix negotiation so that upstream can actually fixate to downstream's rate
4979 If one side has a preference for a particular sample rate or set of sample rates, we
4980 should honour this in the caps we advertise and transform to and from, so that elements
4981 actually know about the other side's sample rate preference and can negotiate to it
4982 if supported. Also add unit test for this.
4984 2009-03-26 19:34:23 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
4986 * gst/playback/gstplaybin2.c:
4987 docs: add a blurb about redirect messages to playbin2 docs
4989 2009-04-01 09:03:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
4991 * gst-libs/gst/rtsp/gstrtspconnection.c:
4992 rtsp: fix little typo in the comments
4994 2009-03-31 17:52:44 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
4996 * gst-libs/gst/rtsp/gstrtspconnection.c:
4997 rtspconnection: make gst_rtsp_watch_queue_message() thread-safe
4998 People might queue messages from a thread other than the thread in which
4999 the main context which this watch is attached is iterated from, so use
5000 a GAsyncQueue instead of a GList, so g_list_append() doesn't trample
5001 over list nodes just freed in the other thread. This just fixes issues
5002 I've had with gst-rtsp-server. We might need more locking in various
5005 2009-03-31 18:13:19 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5007 * gst-libs/gst/rtsp/gstrtspconnection.c:
5008 * gst-libs/gst/rtsp/gstrtspmessage.c:
5009 rtsp: clear the entire builder structure
5010 And use structure instead of variable with sizeof when
5011 clearing the rtsp message structure, for clarity.
5013 2009-03-31 17:56:24 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5015 * gst-libs/gst/rtsp/gstrtspmessage.c:
5016 docs: fix typo in gst_rtsp_message_unset() API docs
5018 2009-03-31 19:00:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5020 * gst-libs/gst/rtsp/gstrtspconnection.c:
5021 * gst-libs/gst/rtsp/gstrtspconnection.h:
5022 rtsp: add support for proxies
5023 Add suport for proxy servers. Currently only used for tunneled HTTP
5024 connections without authentication.
5026 2009-03-31 18:57:08 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
5028 * gst-libs/gst/rtsp/gstrtspmessage.c:
5029 Revert "rtsp: reset whole message (was sizeof pointer instead of sizeof type)"
5030 This reverts commit 79de0b8d67df6fbbe79455adc2e06858295f5c03.
5032 2009-03-26 18:54:56 +0200 Stefan Kost <ensonic@users.sf.net>
5034 * sys/xvimage/xvimagesink.c:
5035 xvimagesink: use xcontext->depth instead of bits in attr.max_value for colorkey
5036 According to the drivers in http://cgit.freedesktop.org/xorg/driver/ we should
5037 format the colorkey depending on xcontext->depth. This is what they will use to
5038 interprete the value. The max_value in turn is usualy a constant regardless of
5041 2009-03-31 12:22:14 +0300 Stefan Kost <ensonic@users.sf.net>
5043 * gst-libs/gst/rtsp/gstrtspmessage.c:
5044 rtsp: reset whole message (was sizeof pointer instead of sizeof type)
5046 2009-03-31 00:56:18 +0100 Jan Schmidt <thaytan@noraisin.net>
5048 * gst-libs/gst/interfaces/mixer.c:
5049 doc: Fix a typo in the GstMixer docs
5051 2009-03-29 12:01:33 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5053 * gst/videoscale/vs_scanline.c:
5054 videoscale: Fix linear scaling for one byte components
5057 2009-03-29 11:53:40 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5059 * gst/videoscale/vs_4tap.c:
5060 videoscale: Fix 4tap scaling of YUYV and friends
5062 2009-03-28 16:08:44 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5064 * gst/videoscale/vs_image.c:
5065 * gst/videoscale/vs_scanline.c:
5066 * gst/videoscale/vs_scanline.h:
5067 videoscale: Rewrite YUYV (and friends) scaling and don't read/write over line ends
5068 Partially fixes bug #577054, there's just one issue left now.
5070 2009-03-28 12:48:04 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5072 * tests/check/elements/videoscale.c:
5073 videoscale: Add some more unit tests
5075 2009-03-28 11:51:01 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5077 * gst/videoscale/gstvideoscale.c:
5078 videoscale: Use bilinear instead of 4tap scaling for heights < 4
5079 Partially fixes bug #577054.
5081 2009-03-28 11:45:41 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5083 * gst/videoscale/vs_scanline.c:
5084 videoscale: Don't read after the end of a line when lineary scaling YUYV/UYVY/RGB/RGBA
5085 This case is for upscaling a frame with width=1
5086 Partially fixes bug #577054.
5088 2009-03-28 11:27:56 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5090 * gst/videoscale/vs_scanline.c:
5091 videoscale: Don't read after the end of a line when lineary scaling YUYV/UYVY
5092 Partially fixes bug #577054.
5094 2009-03-28 10:40:43 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5096 * gst/videotestsrc/gstvideotestsrc.c:
5097 videotestsrc: Initialize buffer memory with zeroes
5098 This prevents valgrind warnings when accessing the "x" parts
5099 of xRGB and friends in other elements that handle (and can handle)
5100 xRGB like ARGB (for example videoscale).
5102 2009-03-28 10:25:12 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5104 * tests/check/Makefile.am:
5105 * tests/check/elements/videoscale.c:
5106 videoscale: Add a lot of unit tests
5108 2009-03-28 10:06:24 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5110 * gst/videoscale/gstvideoscale.c:
5111 videocale: Add support for video/x-raw-gray with bpp=depth=8
5113 2009-03-28 10:01:00 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5115 * gst/videotestsrc/videotestsrc.c:
5116 videotestsrc: Add support for generating video/x-raw-gray with bpp=depth=8
5118 2009-03-28 09:43:23 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5120 * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
5121 ffmpegcolorspace: video/x-raw-gray is the same as the YUV Y800 format
5123 2009-03-27 19:12:49 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5125 * gst/videoscale/vs_4tap.c:
5126 videoscale: Take the next luma value instead of every second next when scaling UYVY and friends
5128 2009-03-27 19:09:47 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5130 * gst/videoscale/gstvideoscale.c:
5131 videoscale: Add support for v308 YUV colorspace
5133 2009-03-27 13:15:11 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5135 * gst/videoscale/vs_4tap.c:
5136 videoscale: Add my copyright to the 4tap scalers
5138 2009-03-27 13:14:17 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5140 * gst/videoscale/gstvideoscale.c:
5141 videoscale: Enable 4-tap scaling for all supported formats
5143 2009-03-27 13:14:00 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5145 * gst/videoscale/vs_4tap.c:
5146 * gst/videoscale/vs_4tap.h:
5147 videoscale: Implement 4-tap scaling for RGB565 and RGB555
5149 2009-03-27 10:47:39 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5151 * gst/videoscale/vs_4tap.c:
5152 * gst/videoscale/vs_4tap.h:
5153 videoscale: Implement 4-tap scaling for UYVY
5155 2009-03-27 09:33:58 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5157 * gst/videoscale/vs_4tap.c:
5158 * gst/videoscale/vs_4tap.h:
5159 videoscale: Implement 4-tap scaling for YUY2 and YVYU
5161 2009-03-26 22:14:53 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5163 * gst/videoscale/vs_4tap.c:
5164 * gst/videoscale/vs_4tap.h:
5165 videoscale: Implement 4-tap scaling for RGB and BGR
5167 2009-03-26 22:08:26 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5169 * gst/videoscale/vs_4tap.c:
5170 * gst/videoscale/vs_4tap.h:
5171 videoscale: Implement 4-tap scaling for RGBA and other 4 byte formats
5173 2009-03-26 11:02:41 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5175 * ext/pango/gsttextoverlay.c:
5176 textoverlay: Fix drawing of UYVY text borders
5178 2009-03-26 10:36:27 +0100 Zeeshan Ali <zeeshan.ali@nokia.com>
5180 * ext/pango/gsttextoverlay.c:
5181 * ext/pango/gsttextoverlay.h:
5182 textoverlay: Add support for UYVY colorspace
5185 2009-03-25 19:01:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5187 * gst/playback/gstdecodebin2.c:
5188 decodebin2: do some more cleanup
5189 Free the groups when we go to READY.
5190 Allow for NO_PREROLL elements.
5192 2009-03-25 16:37:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5194 * gst-libs/gst/rtsp/gstrtspconnection.c:
5195 rtsp: start CSeq counting from 1 instead of 0
5196 Start counting from 1 instead of 0 as this is what most other clients
5199 2009-03-25 16:35:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5201 * gst-libs/gst/rtsp/gstrtspdefs.c:
5202 * gst-libs/gst/rtsp/gstrtspdefs.h:
5203 rtsp: add ETag and If-Match headers
5204 Add new headers, we need them for RealMedia support.
5206 2009-03-25 14:16:25 +0200 Stefan Kost <ensonic@users.sf.net>
5208 * sys/xvimage/xvimagesink.c:
5209 xvimagesink: scale the colorkey components in case of 16bit visuals
5210 Use a default that won't be scales to 0,0,0
5212 2009-03-25 11:27:44 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5214 * gst-libs/gst/audio/gstbaseaudiosrc.c:
5215 audiosrc: improve 'Dropped n samples' warning message
5217 2009-03-24 19:41:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5219 * tests/examples/app/appsrc-ra.c:
5220 * tests/examples/app/appsrc-seekable.c:
5221 examples: use new method to set flags
5222 Use the new core method for setting object enum properties by name.
5224 2009-03-24 18:29:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5226 * gst/playback/gstplaysink.c:
5227 * gst/playback/gstplaysink.h:
5228 playbin2: add more support for subpictures
5230 2009-03-24 17:12:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5232 * gst/playback/gstplaybin2.c:
5233 * gst/playback/gstplaysink.c:
5234 * gst/playback/gstplaysink.h:
5235 playbin2: first support for subpictures
5236 Add beginnings of subpicture support.
5238 2009-03-24 15:26:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5240 * tests/examples/seek/seek.c:
5241 seek: print tags from the different tracks
5243 2009-03-24 12:22:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5245 * gst/playback/gstplaybin2.c:
5246 playbin2: blacklist subpictures for now
5247 Blacklist the subpictures until we add support for them.
5248 Add some small debug info.
5251 2009-03-24 12:19:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5253 * gst/playback/gsturidecodebin.c:
5254 uridecodebin: expose more media types
5255 Expose more media types from a raw source, such as the subpicture and various
5257 Small cleanups and add some more debugging.
5260 2009-03-24 10:42:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5262 * gst/playback/gstplaysink.c:
5263 playbin2: rescan audio sinks for volume/mute
5264 Rescan the audio sinks for the mute and volume properties.
5267 2009-03-23 19:40:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5269 * gst/playback/gstplaysink.c:
5270 playbin2: fix reuse of the video chains
5271 When reusing playbin with visualisations, reset the async property on the video
5272 sink because some sinks might dynamically recreate their sinks.
5275 2009-03-23 17:37:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5277 * gst/playback/gstplaysink.c:
5278 playbin2: allow dynamic swtiching of subtitles
5279 When we have the textpad configured, enable and disable the subtitles by setting
5280 the silent flag on the overlay element instead of trying to remove elements.
5283 2009-03-23 16:59:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5285 * tests/icles/playbin-text.c:
5286 tests: print some more info in the text example
5287 Print both the position and the running_time when the subtitle becomes available
5290 2009-03-23 16:04:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5292 * gst/playback/gstplaysink.c:
5293 playbin2: fix dynamic switching of visualisations
5294 Fix the switching of visualisations by requesting and releasing the tee request
5298 2009-03-23 16:19:11 +0200 Stefan Kost <ensonic@users.sf.net>
5301 * gst/tcp/gsttcpclientsink.c:
5302 * gst/tcp/gsttcpclientsrc.c:
5303 * gst/tcp/gsttcpserversink.c:
5304 * gst/tcp/gsttcpserversrc.c:
5305 docs: add examples for tcp elements, also use correct section name. Fixes #564139
5306 Updated the examples in the README to actually work. Add them to api docs. Tests
5307 the api-docs and fix the section names to make the docs actualy show up.
5308 The example for "tcpserversrc" needs review (might be an element bug).
5310 2009-03-17 09:14:02 +0200 Stefan Kost <ensonic@users.sf.net>
5312 * gst/videoscale/gstvideoscale.c:
5313 indent: fix damange that gst-indent did some time ago
5315 2009-03-23 15:27:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5317 * gst/playback/gstplaysink.c:
5318 playbin2: fix linking order
5319 Link after doing the state change and unlink before shutting down. Makes the
5320 window for causing races in toggling the visualisations smaller.
5323 2009-03-23 12:26:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5325 * gst/playback/gsturidecodebin.c:
5326 uridecodebin: reset counter
5327 reset the number of pending dynamic operations back to 0 when we reuse
5331 2009-03-23 11:38:53 +0100 Edward Hervey <bilboed@bilboed.com>
5333 * ext/theora/theoradec.c:
5334 theoradec: Use GST_CLOCK_TIME_NONE for invalid positions. Fixes #543591
5335 The problem was that previously we didn't check whether _theora_granule_frame
5336 returned a negative framecount or not, resulting in bogus timestamps.
5338 2009-03-21 09:46:28 +0100 René Stadler <mail@renestadler.de>
5340 * ext/vorbis/vorbisenc.c:
5341 vorbisenc: Set caps on non-header ouput buffers.
5344 2009-03-20 16:13:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5346 * tests/examples/seek/seek.c:
5347 seek: Add some more debug
5348 Add some more info about the selected streams.
5350 2009-03-20 15:47:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5352 * gst/playback/gstdecodebin2.c:
5353 decodebin2: a pad starts out being not drained.
5354 Mark a new pad as not drained until we get EOS on it.
5356 2009-03-20 14:17:19 +0100 LRN <lrn1986 at gmail dot com>
5358 * gst/playback/gstqueue2.c:
5359 win32: fix seeking in large files
5360 Fix Seeking in large files by using the 64-bit seek functions.
5363 2009-03-19 20:31:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5365 * gst/playback/gstdecodebin2.c:
5366 decodebin2: recover from failing to add a pad
5367 When we cannot add a pad to the decodebin2 for some reason, print a warning but
5368 continue adding the remaining pads.
5370 2009-03-19 19:35:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5372 * gst/playback/gstdecodebin2.c:
5373 decodebin2: more cleanups and docs.
5374 Add some more comments and use g_list_prepend().
5376 2009-03-19 19:19:38 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5378 * gst/playback/gstdecodebin2.c:
5379 decodebin2: refactoring and race fixes
5380 Refactor some code so that we can take the right locks and in the right order.
5381 Fixes quite a bit of races already.
5383 2009-03-19 19:03:25 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5385 * gst/playback/gstplaybin2.c:
5386 playbin2: remove the group cond + cleanups
5387 Remove the group GCond that we used for waiting for groups to finish because we
5388 use pad blocking on the selectors and counters instead for waiting for the
5390 remove the obsolete about_to_finish variable set while emiting the
5391 about-to-finish signal and fix some old comments.
5392 We don't need to take the playbin lock when querying the uridecodebin.
5394 2009-03-18 10:45:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5396 * tests/icles/playbin-text.c:
5397 icles: print better error and warning messages
5400 2009-03-17 22:53:44 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5402 * gst-libs/gst/rtsp/gstrtspbase64.c:
5403 * gst-libs/gst/rtsp/gstrtspbase64.h:
5404 rtsp: Use GLib base64 functions and deprecate gst_rtsp_base64_encode
5405 This also fixes another instance of CVE-2008-4316.
5407 2009-03-17 19:53:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5409 * ext/ogg/gstoggdemux.c:
5410 oggdemux: report -1 for duration in push mode
5411 In push mode we must return TRUE from the duration query with a value of -1
5412 meaning that we know that we don't know the duration.
5414 2009-03-17 19:09:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5416 * gst/playback/gstdecodebin2.c:
5417 decodebin2: add extra dynamic ref for demuxers
5418 When we make a group connected to a demuxer, keep an extra dynamic refcount for
5419 the group which is only decremented when no_more_pads or a multiqueue overrun is
5420 detected. This way we avoid a race between exposing the group while more dynamic
5421 refs are added from new pads.
5424 2009-03-17 15:39:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5426 * gst/playback/gstplaysink.c:
5427 playbin2: sync state of the sink correctly
5428 Sync the state of the newly added chains to the state of the parent sink element
5429 to avoid lost async-start messages. Fixes cdda:// async-done message storm.
5431 2009-03-17 11:54:40 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5433 * gst/playback/gstplaybin2.c:
5434 playbin2: return NOT_LINKED for unselected streams
5435 When streams are not selected in the selector, return NOT_LINKED so that
5436 upstream elements can skip decoding. Only do this for audio and video pads
5437 because for text streams the overhead is smaller and they could come from
5440 2009-03-17 11:51:58 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5442 * gst/playback/gstplaysink.c:
5443 playbin: set custom text sink properties
5444 Set the custom sink async=FALSE to not make it participate in preroll because we
5445 are dealing with sparse streams.
5446 Try to set sync=TRUE on the custom text sink.
5448 2009-03-17 11:30:58 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5450 * tests/icles/playbin-text.c:
5451 example: use appsink instead of fakesink
5452 Use appsink instead of fakesink to get the subtitles.
5453 Make things more pretty.
5455 2009-03-17 11:24:00 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5457 * tests/icles/.gitignore:
5458 * tests/icles/Makefile.am:
5459 * tests/icles/playbin-text.c:
5460 examples: add example of intercepting subtitles
5461 Add an example of how to install a custom sink for receiving subtitles in
5464 2009-03-17 11:03:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5466 * tests/check/elements/appsink.c:
5467 tests: fix include in the appsink test
5468 Fix dist by doing the right include.
5470 2009-03-16 16:42:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5472 * gst/playback/gstplaybin2.c:
5473 playbin2: don't try to set invalid stream numbers
5474 Fix a problem with setting the stream numbers because we check for the wrong
5478 2009-03-16 16:16:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5480 * gst/playback/gstplaybin2.c:
5481 playbin2: release the shutdown lock
5482 Release the shutdown lock when we wait for other groups to complete or else we
5483 have a deadlock when the other group completes and tries to grab the shutdown
5487 2009-03-16 15:31:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5489 * tests/examples/app/appsrc-ra.c:
5490 * tests/examples/app/appsrc-seekable.c:
5491 * tests/examples/app/appsrc-stream.c:
5492 * tests/examples/app/appsrc-stream2.c:
5493 examples: fix g_object_set() value type.
5494 Make sure we cast the length value as a gint64 to the vararg g_object_set() just
5495 incase sizeof(gsize) != sizeof(gint64).
5497 2009-03-15 19:57:36 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5499 * gst/typefind/gsttypefindfunctions.c:
5500 typefinding: make flac typefinder return lower probability for frame headers
5501 The flac frame header typefinder overstates the likelihood of a match, leading
5502 to false positives with e.g. aac streams and PDF files. Reduce probabilty
5503 returned from LIKELY to POSSIBLE for the frame header matchin code.
5506 2009-03-11 12:59:05 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5508 * gst/typefind/gsttypefindfunctions.c:
5509 typefinding: improve image/bmp typefinder
5510 Detect more variations and also bail out in more cases where the values
5511 don't make sense. Furthermore, add width/height and bpp to the caps,
5514 2009-03-13 15:22:42 +0000 Jan Schmidt <thaytan@noraisin.net>
5516 * tests/check/Makefile.am:
5517 check: Ignore alsamixer in the states test too
5519 2009-03-13 15:22:11 +0000 Jan Schmidt <thaytan@noraisin.net>
5521 * sys/v4l/v4l_calls.c:
5522 v4lsrc: Fix some valgrind warnings about leaked memory and uninitialised data.
5524 2009-03-13 16:19:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5526 * gst-libs/gst/rtsp/gstrtspconnection.c:
5527 rtsp: fix resolving of hostnames
5528 We were returning a pointer to a stack variable with the resolved hostname,
5530 return a copy of the resolved ip address instead.
5533 2009-03-13 15:29:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5535 * ext/vorbis/vorbisparse.c:
5536 vorbisparse: be smarter when queueing headers
5537 Look at the first buffer byte to see if a buffer is a header instead of counting
5540 2009-03-13 15:27:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5542 * ext/theora/gsttheoraparse.h:
5543 * ext/theora/theoraparse.c:
5544 theoraparse: be smarter when queuing headers
5545 Look at the first byte of the buffer data (if we can) to decide if the packet is
5546 a header packet or not instead of counting packets.
5548 2009-03-13 15:26:40 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5550 * ext/ogg/gstoggdemux.c:
5551 oggdemux: add some debug info
5552 Add some debug info to log when the seek worked.
5554 2009-03-13 15:14:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5556 * gst-libs/gst/app/gstappsrc.c:
5557 appsrc: release lock in _eos flushing case
5558 Release the mutex when we are flushing in gst_app_src_end_of_stream()
5561 2009-03-13 11:49:10 +0000 Jan Schmidt <thaytan@noraisin.net>
5563 * ext/vorbis/vorbisdec.c:
5564 vorbisdec: Avoid an unnecessary memory allocation in vorbiscomment handling.
5566 2009-03-13 11:48:28 +0000 Jan Schmidt <thaytan@noraisin.net>
5568 * ext/theora/theoradec.c:
5569 theoradec: Avoid an unnecessary memory allocation in vorbiscomment handling.
5571 2009-03-12 18:27:25 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5573 * gst/playback/gsturidecodebin.c:
5574 playbin2: fix raw elements like cdda://
5575 Fix a fixme with a one liner and make cd playback work again.
5577 2009-03-12 17:47:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5579 * gst/playback/gstplaybin2.c:
5580 * gst/playback/gstplaysink.c:
5581 * gst/playback/gstplaysink.h:
5582 playbin2: improve subtitle handling
5583 Add property to playbin2 to configure a custom sink that receives the raw
5584 subtitle buffers instead of using a textoverlay.
5585 Improve the property finding code to make it more usable.
5586 Use property find code to find async properties in custom sinks that are bins.
5587 Improve text overlay code to gracefully handle missing elements.
5589 2009-02-24 15:58:42 +0000 Jan Schmidt <thaytan@noraisin.net>
5591 * gst-libs/gst/tag/gstvorbistag.c:
5592 vorbistag: Protect memory allocation calculation from overflow.
5593 Patch by: Tomas Hoger <thoger@redhat.com> Fixes CVE-2009-0586
5595 2009-03-12 11:34:20 +0000 Jan Urbanski <jurbanski@flumotion.com>
5597 * gst-plugins-base.spec.in:
5600 2009-03-11 18:45:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5602 * gst-libs/gst/rtsp/gstrtspconnection.c:
5603 rtsp: fix parsing of the timeout parameter
5606 2009-03-11 16:20:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5608 * gst-libs/gst/rtsp/gstrtspmessage.c:
5609 rtsp: fix g_return condition
5610 when parsing a data message, we require a data message.
5612 2009-03-11 13:33:33 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5614 * gst/typefind/gsttypefindfunctions.c:
5615 typefinding: flac typefinder fixes
5616 Use scan context for initial peek as well. Peek 6 bytes in the initial
5617 peek rather than 5 bytes, to match the length of the memcmp we're doing
5618 on that data later. Return immediately when we found caps from looking
5619 at the beginning of the data - no point in continuing to scan the next
5620 64kB for something matching a frame header.
5622 2009-03-11 14:08:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5624 * gst-libs/gst/rtsp/gstrtspmessage.c:
5625 rtsp: free the right string.
5626 Free the key value before we remove the header item from the array. The item we
5627 retrieved from the array is only valid until we remove it from the array.
5629 2009-03-11 14:07:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5631 * gst-libs/gst/rtsp/gstrtspconnection.c:
5632 rtsp: keep track of amount of decoded bytes
5633 Keep track of the actual amount of decoded bytes, which can be less than 3 when
5634 we decode the last bits of a base64 message.
5636 2009-03-10 21:00:26 +0200 Stefan Kost <ensonic@users.sf.net>
5638 * gst/adder/gstadder.c:
5639 adder: log details in getcaps like in setcaps
5641 2009-03-10 13:11:09 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5644 win32: update MANIFEST, fixing 'make dist'
5646 2009-03-09 23:12:00 +0000 Jan Schmidt <thaytan@noraisin.net>
5649 Automatic update of common submodule
5650 From 7032163 to f8b3d91
5652 2009-03-09 16:19:40 +0100 Jonathan Matthew <notverysmart at gmail dot com>
5654 * gst/typefind/gsttypefindfunctions.c:
5655 typefind: add photoshop typefind functions
5656 Add photoshop typefind functions.
5659 2009-03-09 15:46:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5661 * gst/playback/gstdecodebin2.c:
5662 decodebin2: only remove pads that were added
5663 Flag pads that were added so that we can see if we need to remove them later or
5666 2009-03-09 13:53:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5668 * gst-libs/gst/rtsp/gstrtsptransport.c:
5669 rtsp: only add ports when not using TCP
5670 Only add the port numbers in the transport string when we are using udp or
5673 2009-03-09 13:53:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5675 * gst-libs/gst/rtsp/gstrtspmessage.c:
5676 rtsp: use gstreamer dump mem
5679 2009-03-09 13:51:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5681 * gst-libs/gst/rtsp/gstrtspconnection.c:
5682 rtsp: use glib base64 encoder
5685 2009-03-06 19:28:37 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
5687 * gst/playback/gstdecodebin2.c:
5688 Unblock blocked ghostpads when shutting down. Fixes #574293.
5690 2009-03-09 10:03:13 +0100 Edward Hervey <bilboed@bilboed.com>
5692 * gst-libs/gst/riff/riff-media.c:
5693 Riff: Add mapping for Fraps video codec.
5694 Found through insanity testrun. Confirmed mapping in libavformat.
5696 2009-03-09 09:07:13 +0100 Edward Hervey <bilboed@bilboed.com>
5698 * gst-libs/gst/riff/riff-media.c:
5699 riff: Add the 'DVR ' mapping for mpeg2video.
5700 Found this in 3 files from the insanity suite and mapping is also present
5703 2009-03-09 09:06:40 +0100 Edward Hervey <bilboed@bilboed.com>
5705 * gst/typefind/gsttypefindfunctions.c:
5706 typefind: Use the proper data pointer instead of poking random memory.
5708 2009-03-08 18:17:48 +0100 LRN <lrn1986@gmail.com>
5710 * gst-libs/gst/rtsp/gstrtspconnection.c:
5711 rtsp: fix compilation on windows.
5712 Remove unused variable when building for windows.
5715 2009-03-08 12:03:22 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5718 Automatic update of common submodule
5719 From ffa738d to 7032163
5721 2009-03-08 11:19:00 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5724 Automatic update of common submodule
5725 From 3f13e4e to ffa738d
5727 2009-03-07 11:44:40 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5730 Automatic update of common submodule
5731 From 3c7456b to 3f13e4e
5733 2009-03-07 10:44:43 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
5736 Automatic update of common submodule
5737 From 57c83f2 to 3c7456b
5739 2009-03-06 19:02:58 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5741 * ext/theora/theoradec.c:
5742 theoradec: parse and use codec_data in the caps
5743 Parse the codec_data in the caps and use this as the headers.
5746 2009-03-06 18:53:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5748 * gst-libs/gst/riff/riff-media.c:
5749 riff: add theora mapping
5750 Add theora mappings. See #574169.
5752 2009-03-06 16:31:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5754 * gst-libs/gst/rtsp/gstrtspconnection.c:
5755 * gst-libs/gst/rtsp/gstrtspconnection.h:
5756 * win32/common/libgstrtsp.def:
5757 rtsp: Add methods for getting the read/write fds
5758 API:gst_rtsp_connection_get_readfd()
5759 API:gst_rtsp_connection_get_writefd()
5761 2009-03-06 10:35:01 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5764 * win32/common/audio-enumtypes.c:
5765 win32: indent copied *-enumtypes.c files in make win32-update
5767 2009-03-06 10:35:56 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5770 win32: update MANIFEST
5772 2009-03-06 10:30:28 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5775 * win32/common/config.h:
5776 win32: fix configure logic for GST_INSTALL_PLUGINS_HELPER define
5778 2009-03-06 10:05:11 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5780 * win32/common/_stdint.h:
5781 * win32/common/config.h:
5782 * win32/common/gstrtsp-enumtypes.c:
5783 * win32/common/interfaces-enumtypes.c:
5784 * win32/common/multichannel-enumtypes.c:
5785 * win32/common/pbutils-enumtypes.c:
5786 * win32/common/video-enumtypes.c:
5787 * win32/common/video-enumtypes.h:
5788 win32: update windows files via make win32-update
5789 Updates win32 files using the new system/hook, and defines HAVE_PROCESS_H,
5790 which fixes the build of pbutils on windows (#574319).
5792 2009-03-06 10:03:31 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5795 gitignore: ignore more
5797 2009-03-06 10:37:38 +0100 Julien Moutte <julien@fluendo.com>
5799 * gst-libs/gst/rtsp/gstrtspconnection.c:
5800 Fix build on Mac OS X
5802 2009-03-05 15:42:23 -0800 Michael Smith <msmith@songbirdnest.com>
5804 * gst/playback/gstdecodebin2.c:
5805 decodebin2: don't stay connected to notify::caps after negotiation
5806 Disconnect the notify::caps signal in our callback (it'll be re-added
5807 if we're not, in fact, finished getting complete caps). Ensures that
5808 caps changes mid-stream (e.g. from an mp3 that changes from
5809 stereo->mono mid-file) don't cause us to try to add a new pad.
5811 2009-03-05 13:48:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5813 * gst-libs/gst/rtsp/gstrtsprange.c:
5814 rtsp: fix parsing of 'now-' ranges.
5817 2009-03-05 12:43:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5819 * tests/examples/dynamic/.gitignore:
5820 * tests/examples/dynamic/Makefile.am:
5821 * tests/examples/dynamic/sprinkle.c:
5822 * tests/examples/dynamic/sprinkle2.c:
5823 * tests/examples/dynamic/sprinkle3.c:
5824 examples: add some more sprinkle examples
5825 Add some more sprinle examples and add some more comments.
5828 2009-03-05 11:57:40 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5830 * docs/plugins/gst-plugins-base-plugins-sections.txt:
5831 docs: add appsrc symbols to standard section
5834 2009-03-05 12:27:16 +0200 Stefan Kost <ensonic@users.sf.net>
5836 * gst/adder/gstadder.c:
5837 adder: add variants for unsigned to fix warnings for unneeded check
5838 For unsigned int out+in can't be < 0.
5840 2009-03-05 10:58:12 +0200 Stefan Kost <ensonic@users.sf.net>
5842 * gst/subparse/gstsubparse.c:
5843 subparse: use the right variable in debug log, encoding is not yet initialized
5845 2009-03-05 10:51:25 +0200 Stefan Kost <ensonic@users.sf.net>
5847 * sys/v4l/v4l_calls.c:
5848 v4l: add a fixme for broken code, that someone who has a v4l tuner device should fix
5850 2009-03-05 10:39:33 +0200 Stefan Kost <ensonic@users.sf.net>
5852 * gst/audioresample/gstaudioresample.c:
5853 audioresample: add missing break in event handling, remove dead code
5855 2009-03-04 16:24:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5857 * gst-libs/gst/rtsp/gstrtspconnection.c:
5858 rtsp: do some more cleanup in _close
5859 Do som more cleanup in gst_rtsp_connection_close() so that it's back into the
5860 unconnected state as it was allocated.
5862 2009-03-04 16:11:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5864 * gst-libs/gst/rtsp/gstrtspconnection.c:
5865 * gst-libs/gst/rtsp/gstrtspconnection.h:
5866 rtsp: fix the memory management of the url
5867 Constify the url parameter in _create.
5868 Make a copy of the url stored in the connection.
5869 Free the url when the connection is freed.
5871 2009-03-04 12:21:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5873 * docs/libs/gst-plugins-base-libs-sections.txt:
5874 * gst-libs/gst/rtsp/gstrtspconnection.c:
5875 * gst-libs/gst/rtsp/gstrtspconnection.h:
5876 * win32/common/libgstrtsp.def:
5877 RTSP: Add support for server tunneling
5878 Save the tunnelid in the connection. Add a method to retrieve the tunnelid so
5879 that a server can store and match the id against other tunnel requests.
5880 Fix the URI in the tunnel requests so that they contain the absolute uri and the
5881 query string if any instead of just the hostname.
5882 Transparently base64 decode the input stream when tunneling.
5883 Add method to set the connection ip address so that it can be included in the
5885 Add method to connect the two tunnel requests.
5886 Add two callbacks for the async mode to notify a tunnel start and tunnel
5888 Add method to reset the watch after the connection has been tunneled.
5889 Various little refactoring to make more stuff reusable.
5890 API: RTSP::gst_rtsp_connection_set_ip()
5891 API: RTSP::gst_rtsp_connection_get_tunnelid()
5892 API: RTSP::gst_rtsp_connection_do_tunnel()
5893 API: RTSP::gst_rtsp_watch_reset()
5895 2009-03-04 12:18:00 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5897 * gst-libs/gst/rtsp/gstrtspdefs.c:
5898 * gst-libs/gst/rtsp/gstrtspdefs.h:
5899 rtsp: add new defines for tunneling
5900 Add two more result codes for tunneling support.
5902 2009-03-04 12:12:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5904 * gst-libs/gst/rtsp/gstrtspmessage.h:
5905 rtsp: remove , from last enum member
5906 Remove , from last enum member to improve compatibility with other compilers.
5908 2009-02-28 15:23:20 -0800 LRN <lrn1986@gmail.com>
5910 * gst/subparse/gstsubparse.c:
5911 subparse: Convert regex code to GRegex code
5912 Fixes: #572993. Patch author prefers to use an alias, contact
5913 ds if you actually need a real name.
5914 Signed-off-by: David Schleef <ds@schleef.org>
5916 2009-03-02 16:13:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5918 * gst-libs/gst/rtsp/gstrtspconnection.c:
5919 rtsp: remove debugging g_message
5922 2009-03-02 16:03:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5924 * docs/libs/gst-plugins-base-libs-sections.txt:
5925 * gst-libs/gst/rtsp/gstrtspconnection.c:
5926 * gst-libs/gst/rtsp/gstrtspconnection.h:
5927 * win32/common/libgstrtsp.def:
5928 RTSP: add support for Quicktime tunneled RTSP
5929 Add support for tunneling RTSP over HTTP.
5930 Fix documentation some more.
5932 API: RTSP:gst_rtsp_connection_is_tunneled()
5933 API: RTSP:gst_rtsp_connection_set_tunneled()
5935 2009-03-02 15:48:56 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5937 * gst-libs/gst/rtsp/gstrtsptransport.h:
5938 * gst-libs/gst/rtsp/gstrtspurl.c:
5939 RTSP: parse rtsph uris as RTSP tunneled over HTTP
5940 Add transport define for RTSP tunneled over HTTP.
5941 Parse rtsph:// uris as tunneled HTTP over TCP.
5942 API: GstRTSPLowerTrans::GST_RTSP_LOWER_TRANS_HTTP
5945 2009-03-02 12:48:18 +0100 Edward Hervey <bilboed@bilboed.com>
5947 * win32/common/libgstrtsp.def:
5948 win32: Add gst_rtsp_connection_get_url definition
5949 No, I'm not wim's buildslave, seriously.
5951 2009-03-02 10:58:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
5953 * gst-libs/gst/rtsp/gstrtspconnection.c:
5954 * gst-libs/gst/rtsp/gstrtspconnection.h:
5955 rtsp: add _get_url method and separate sockets
5956 Add gst_rtsp_connection_get_url() method.
5957 Reserve space for 2 sockets, one for reading and one for writing. Use socket
5958 pointers to select the read and write sockets. This should allow us to implement
5959 tunneling over HTTP soon.
5960 API: RTSP::gst_rtsp_connection_get_url()
5962 2009-03-01 18:31:17 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
5964 * gst-libs/gst/app/gstapp-marshal.list:
5965 app: force automatic rebuild of gstapp-marshal.[ch] after previous change
5966 The previous change to appsrc/appsink requires people to 'make clean'
5967 to get the marshallers rebuilt (causing a build failure otherwise).
5968 Change some lines in the .list file around to force a rebuild of
5969 these files automatically.
5971 2009-02-28 11:07:04 -0800 David Schleef <ds@schleef.org>
5974 Bump glib requirement to 2.14
5976 2009-02-28 19:37:53 +0100 LRN <lrn1986@gmail.com>
5978 * ext/gio/gstgiobasesink.c:
5979 gio: Use correct format modifier for size_t
5982 2009-02-28 19:35:33 +0100 LRN <lrn1986@gmail.com>
5984 * gst-libs/gst/rtsp/gstrtspconnection.c:
5985 rtspconnection: Use correct types for some functions on Win32
5988 2009-02-28 13:11:59 +0100 Edward Hervey <bilboed@bilboed.com>
5990 * gst-libs/gst/rtsp/gstrtspconnection.c:
5991 rtspconnection: Fix warning about using unitialized value.
5993 2009-02-28 12:41:28 +0100 Edward Hervey <bilboed@bilboed.com>
5995 * gst-libs/gst/riff/riff-ids.h:
5996 * gst-libs/gst/riff/riff-media.c:
5997 riff: Add more codec mappings.
5998 This comes mostly from a review of ffmpeg/libavformat/riff.c
6000 2009-02-27 11:14:25 +0200 Stefan Kost <ensonic@users.sf.net>
6002 * ext/alsa/gstalsa.c:
6003 alsa: release pcminfo after the strdup
6005 2009-02-26 17:38:47 +0200 Stefan Kost <ensonic@users.sf.net>
6007 * gst-libs/gst/rtsp/gstrtsprange.c:
6008 rtsprange: don't leak the range in case of parsing error.
6009 Free the gstRTSPTimeRange if we don't return it. Also simplify
6010 gst_rtsp_range_free() as it is valid to pass NULL to g_free().
6012 2009-02-26 16:47:39 +0200 Stefan Kost <ensonic@users.sf.net>
6014 * ext/alsa/gstalsa.c:
6015 alsa: cleanup name lookup.
6016 We can break, once we have a name to make sure, we won't read it ever twice.
6018 2009-02-26 16:09:03 +0200 Stefan Kost <ensonic@users.sf.net>
6020 * gst/subparse/gstsubparse.c:
6021 subparse: don't leak line, if flushing
6023 2009-02-26 16:03:39 +0200 Stefan Kost <ensonic@users.sf.net>
6025 * ext/gio/gstgiosink.c:
6026 giosink: reflow error handling to not leak uri
6028 2009-02-26 15:53:10 +0200 Stefan Kost <ensonic@users.sf.net>
6030 * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
6031 * gst/ffmpegcolorspace/imgconvert.c:
6032 ffmpegcolorspace: remove unused code/variables
6034 2009-02-26 12:10:47 +0200 Stefan Kost <ensonic@users.sf.net>
6036 * sys/ximage/ximagesink.c:
6037 ximagesink: use GST_FLOW_NOT_NEGOTIATED for partial caps
6039 2009-02-26 16:44:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6041 * docs/libs/gst-plugins-base-libs-sections.txt:
6042 * gst-libs/gst/app/gstappsink.c:
6043 * gst-libs/gst/app/gstappsrc.c:
6044 * gst-libs/gst/app/gstappsrc.h:
6045 * win32/common/libgstapp.def:
6046 app: add callbacks to appsrc, cleanups
6047 Add a uri handler to appsink.
6048 don't emit signals when we have installed callbacks on appsink.
6049 Add callbacks to appsrc to replace the signals.
6050 Add property to disable callbacks in appsrc, default to TRUE for backwards
6051 compatibility but disable when callbacks are installed.
6052 API: GstAppSrc::emit-signals
6053 API: GstAppSrc::gst_app_src_set_emit_signals()
6054 API: GstAppSrc::gst_app_src_get_emit_signals()
6055 API: GstAppSrc::gst_app_src_set_callbacks()
6057 2009-02-26 11:42:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6059 * docs/libs/gst-plugins-base-libs-sections.txt:
6060 * gst-libs/gst/app/gstappsink.h:
6061 * tests/check/elements/appsink.c:
6062 Appsink: add padding for callbacks + docs
6063 Add some padding to the callbacks structure just to be safe.
6064 Remove the now invisible marshaller methods from the docs.
6065 Fix a comment in the unit test.
6067 2009-02-26 09:52:59 +0100 Edward Hervey <bilboed@bilboed.com>
6069 * win32/common/libgstapp.def:
6070 win32: Add new libgstapp symbol
6072 2009-02-26 10:07:21 +0200 Stefan Kost <ensonic@users.sf.net>
6074 * docs/plugins/gst-plugins-base-plugins-sections.txt:
6075 docs: clean section.txt file.
6076 Add appsrc/sink symbols to private, as they are covered in the libs docs.
6078 2009-02-26 10:06:23 +0200 Stefan Kost <ensonic@users.sf.net>
6080 * gst/playback/gstplaybasebin.c:
6081 docs: fix random text after since: tag. Also fix class name to make the docs actual appear.
6083 2009-02-26 09:56:16 +0200 Stefan Kost <ensonic@users.sf.net>
6085 * docs/plugins/gst-plugins-base-plugins.args:
6086 * docs/plugins/gst-plugins-base-plugins.hierarchy:
6087 * docs/plugins/gst-plugins-base-plugins.interfaces:
6088 * docs/plugins/gst-plugins-base-plugins.prerequisites:
6089 * docs/plugins/inspect/plugin-adder.xml:
6090 * docs/plugins/inspect/plugin-alsa.xml:
6091 * docs/plugins/inspect/plugin-app.xml:
6092 * docs/plugins/inspect/plugin-audioconvert.xml:
6093 * docs/plugins/inspect/plugin-audiorate.xml:
6094 * docs/plugins/inspect/plugin-audioresample.xml:
6095 * docs/plugins/inspect/plugin-audiotestsrc.xml:
6096 * docs/plugins/inspect/plugin-cdparanoia.xml:
6097 * docs/plugins/inspect/plugin-decodebin.xml:
6098 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
6099 * docs/plugins/inspect/plugin-gdp.xml:
6100 * docs/plugins/inspect/plugin-gio.xml:
6101 * docs/plugins/inspect/plugin-gnomevfs.xml:
6102 * docs/plugins/inspect/plugin-libvisual.xml:
6103 * docs/plugins/inspect/plugin-ogg.xml:
6104 * docs/plugins/inspect/plugin-pango.xml:
6105 * docs/plugins/inspect/plugin-playback.xml:
6106 * docs/plugins/inspect/plugin-queue2.xml:
6107 * docs/plugins/inspect/plugin-subparse.xml:
6108 * docs/plugins/inspect/plugin-tcp.xml:
6109 * docs/plugins/inspect/plugin-theora.xml:
6110 * docs/plugins/inspect/plugin-typefindfunctions.xml:
6111 * docs/plugins/inspect/plugin-uridecodebin.xml:
6112 * docs/plugins/inspect/plugin-video4linux.xml:
6113 * docs/plugins/inspect/plugin-videorate.xml:
6114 * docs/plugins/inspect/plugin-videoscale.xml:
6115 * docs/plugins/inspect/plugin-videotestsrc.xml:
6116 * docs/plugins/inspect/plugin-volume.xml:
6117 * docs/plugins/inspect/plugin-vorbis.xml:
6118 * docs/plugins/inspect/plugin-ximagesink.xml:
6119 * docs/plugins/inspect/plugin-xvimagesink.xml:
6120 * gst/playback/gstplaybin2.c:
6121 docs: playbin2 has no stream-info
6123 2009-02-26 09:53:03 +0200 Stefan Kost <ensonic@users.sf.net>
6125 * gst-libs/gst/video/video.h:
6126 docs: fix newly added interlace constants and plug holes in video format docs
6128 2009-02-26 09:35:43 +0200 Stefan Kost <ensonic@users.sf.net>
6130 * gst-libs/gst/app/gstappsink.c:
6131 * gst-libs/gst/app/gstappsrc.c:
6132 * gst-libs/gst/audio/gstaudiofilter.c:
6133 * gst-libs/gst/audio/gstringbuffer.c:
6134 * gst-libs/gst/rtp/gstrtcpbuffer.c:
6135 docs: don't put random stuff in tags.
6136 Tags like Since: or Returns: can only be followed by more tags. gtk-doc has no
6137 tag to append text again to the documentation body.
6139 2009-02-06 11:10:15 +0200 Stefan Kost <ensonic@users.sf.net>
6141 * sys/ximage/ximagesink.c:
6142 ximagsink: do not access uninitialized height variable.
6143 Exit like in xvimagesink, if we have partial caps.
6145 2009-02-25 20:26:05 -0800 David Schleef <ds@schleef.org>
6149 * win32/common/config.h.in:
6150 Change how win32/common/config.h is updated
6151 Generate win32/common/config.h-new directly from config.h.in,
6152 using shell variables in configure and some hard-coded information.
6153 Change top-level makefile so that 'make win32-update' copies the
6154 generated file to win32/common/config.h, which we keep in source
6155 control. It's kept in source control so that the git tree is
6157 This change is similar to the one recently applied to GStreamer,
6158 except that it adds a few -base specific defines.
6160 2009-02-25 19:40:43 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6162 * gst-libs/gst/app/Makefile.am:
6163 * gst-libs/gst/app/gstappsink.c:
6164 * gst-libs/gst/app/gstappsrc.c:
6165 * win32/common/libgstapp.def:
6166 app: add win32 .def file and only export functions we want exported
6167 Add a .def file for win32 builds (and make check-exports).
6168 Fix LDFLAGS in Makefile.am, so the usual export regexps are used (fixes #573165).
6169 Make sure private marshaller functions aren't exported by prefixing them with __gst;
6170 also rename gst_app_marshal_OBJECT__VOID to _BUFFER__VOID, make it static and add
6171 a comment why we're not using glib-genmarshal for this one.
6173 2009-02-25 17:08:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6175 * tests/examples/dynamic/.gitignore:
6176 * tests/examples/dynamic/Makefile.am:
6177 * tests/examples/dynamic/sprinkle.c:
6178 sprinkle: Add another example app
6179 Add an example app that dynamically adds and removes audiotestsrc elements from
6182 2009-02-25 16:25:33 +0100 Peter Kjellerstedt <pkj@axis.com>
6184 * gst-libs/gst/rtsp/gstrtspconnection.c:
6187 2009-02-25 11:31:02 +0100 Peter Kjellerstedt <pkj@axis.com>
6189 * gst-libs/gst/rtsp/gstrtspconnection.c:
6190 * gst/tcp/gstmultifdsink.c:
6191 rtsp, multifdsink: Unify the use of union gst_sockaddr.
6193 2009-02-25 14:22:35 +0000 Jan Schmidt <thaytan@noraisin.net>
6197 build: Update shave init statement for changes in common. Bump common.
6199 2009-02-25 13:16:32 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6201 * sys/xvimage/xvimagesink.c:
6202 * sys/xvimage/xvimagesink.h:
6203 xvimageink: protect buffer_alloc from shutdown
6204 Use the pool_lock in the buffer_alloc function to detect shutdown. Avoids
6205 crashes when the sink is shutdown.
6207 2009-02-25 12:43:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6209 * gst/playback/gstplaybin2.c:
6210 playbin: use flushing pads instead of fakesink
6211 Use the flushing pads on playsink to terminate on shutdown instead of plugging
6212 fakesinks. this should be a little cheaper.
6214 2009-02-25 12:42:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6216 * gst/playback/gstplaysink.c:
6217 * gst/playback/gstplaysink.h:
6218 playsink: Add FLUSHING pad type
6219 Make it possible to request a flushing pad from the playsink. We can eventually
6220 use these flushing pads to quickly terminate the dataflow when we are shutting
6223 2009-02-25 11:31:52 +0000 Jan Schmidt <thaytan@noraisin.net>
6226 Automatic update of common submodule
6227 From 9cf8c9b to a6ce5c6
6229 2009-02-25 09:52:38 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6231 * gst-libs/gst/riff/riff-media.c:
6232 riff: add fourcc for mpeg2-in-avi (as produced by mencoder)
6235 2009-02-25 12:07:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6237 * tests/icles/stress-playbin.c:
6238 stress-playbin: print the current uri
6239 Print the current uri so that we can more easily see what uri caused a crash or
6242 2009-02-25 11:07:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6244 * tests/icles/stress-playbin.c:
6245 Print the errors more clearly
6246 Print some more verbose messages when dealing with errors.
6248 2009-02-25 10:08:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6250 * gst/playback/gstplaybin2.c:
6251 Release the group lock when setting states
6252 Release the group lock while we perform the state changes on the uridecodebins
6253 because that might trigger callbacks that we need to handle with the group lock
6254 taken. Avoids a possible deadly embrace in some id3/flac files.
6257 2009-02-25 10:05:38 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6259 * gst/playback/gstdecodebin2.c:
6260 Combine finding and creating groups
6261 Combine the search for the current group and optionally creating one into one
6262 function so that we can avoid taking the lock multiple times.
6264 2009-02-25 08:22:00 +0100 Edward Hervey <bilboed@bilboed.com>
6266 * gst/playback/gstplaybin2.c:
6267 Playbin2: Don't leave unused parameters in debug statements.
6268 Fixes build on macosx
6270 2009-02-24 10:33:05 +0100 Edward Hervey <bilboed@bilboed.com>
6272 * gst-libs/gst/riff/riff-media.c:
6273 Riff: Add fourcc for mpeg1-in-avi (as produced by mencoder)
6275 2009-02-24 18:43:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6277 * gst/playback/gstplaybin2.c:
6278 Add some G_UNLIKELY because we can
6279 Add a G_UNLIKELY when checking the shutdown variable.
6281 2009-02-24 17:23:58 +0000 Garret D'Amore <garrett.damore@sun.com>
6283 * gst-libs/gst/interfaces/mixer.h:
6284 * gst-libs/gst/interfaces/mixertrack.h:
6285 mixer interface: Add flags to enhance mixer interfaces
6286 This patch adds a few flags to the mixer and mixerctrl interface to
6287 better support OSSv4 (and potentially other backends).
6288 Patch By: Garret D'Amore <garrett.damore@sun.com>
6289 Signed-Off-By: Jan Schmidt <jan.schmidt@sun.com>
6290 API: GST_MIXER_FLAG_HAS_WHITELIST, GST_MIXER_FLAG_GROUPING,
6291 API: GST_MIXER_TRACK_NO_RECORD, GST_MIXER_TRACK_NO_MUTE,
6292 API: GST_MIXER_TRACK_WHITELIST
6294 2009-02-24 17:03:08 +0000 Jan Schmidt <thaytan@noraisin.net>
6296 * gst/tcp/gstmultifdsink.c:
6297 multifdsink: Fix strict aliasing error using a union
6299 2009-02-24 16:49:40 +0000 Jan Schmidt <thaytan@noraisin.net>
6301 * gst-libs/gst/rtsp/gstrtspconnection.c:
6302 rtsp: Fix a strict aliasing warning
6303 Fix strict aliasing warnings from casting a sockaddr_storage and
6304 using it as a sockaddr_in6. Use a union instead.
6306 2009-02-24 16:08:49 +0000 Jan Schmidt <thaytan@noraisin.net>
6308 * docs/libs/.gitignore:
6309 * docs/libs/tmpl/.gitignore:
6310 * docs/plugins/.gitignore:
6311 * docs/plugins/tmpl/.gitignore:
6312 Remove .gitignore files from the docs tmpl dirs, that are killed by make clean.
6314 2009-02-24 14:36:39 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6316 * docs/plugins/Makefile.am:
6317 * ext/vorbis/Makefile.am:
6318 * ext/vorbis/gstvorbisdec.h:
6319 * ext/vorbis/gstvorbisenc.h:
6320 * ext/vorbis/gstvorbisparse.h:
6321 * ext/vorbis/gstvorbistag.h:
6322 * ext/vorbis/vorbis.c:
6323 * ext/vorbis/vorbisdec.c:
6324 * ext/vorbis/vorbisdec.h:
6325 * ext/vorbis/vorbisenc.c:
6326 * ext/vorbis/vorbisenc.h:
6327 * ext/vorbis/vorbisparse.c:
6328 * ext/vorbis/vorbisparse.h:
6329 * ext/vorbis/vorbistag.c:
6330 * ext/vorbis/vorbistag.h:
6331 vorbis: Rename vorbis*.h to gstvorbis*.h to prevent name conflicts
6333 2009-02-24 14:06:38 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6335 * gst/ffmpegcolorspace/avcodec.h:
6336 * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
6337 * gst/ffmpegcolorspace/imgconvert.c:
6338 ffmpegcolorspace: Add conversion from/to YVYU colorspace
6341 2009-02-24 13:42:01 +0100 Jonas Danielsson <jonas.danielsson@axis.com>
6343 * gst/ffmpegcolorspace/imgconvert.c:
6344 ffmpegcolorspace: Add direct UYVY->GRAY8 conversion
6345 The conversion from UYVY to RGB24 and then to GRAY8
6346 is quite slow. Fixes bug #569655.
6348 2009-02-19 17:16:51 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
6350 * gst/playback/gstplaybin2.c:
6351 playbin2: fix deadlock when shutting down. Fixes #572577.
6353 2009-02-19 17:15:18 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
6355 * tests/icles/stress-playbin.c:
6356 stress-playbin: make more flexible, e.g. also useful for playbin2
6358 2009-02-24 12:11:00 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6360 * gst-libs/gst/rtsp/gstrtspconnection.c:
6361 Match WSAStartup and WSACleanup correctly
6362 Don't randomly call WSAStartup and WSACleanup but instead call the startup when
6363 we create a connection and cleanup when we free it again. Because the internal
6364 datastructure is refcounted, this should not cause any refcounting leaks when
6365 the connection is managed correctly.
6368 2009-02-18 11:59:58 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
6370 * gst/playback/gstplaysink.c:
6371 playbin2/playsink: Set audiotee to PAUSED state in all cases. Fixes #565105.
6373 2009-02-23 10:57:42 -0800 David Flynn <davidf@rd.bbc.co.uk>
6375 * pkgconfig/gstreamer-app-uninstalled.pc.in:
6376 * pkgconfig/gstreamer-audio-uninstalled.pc.in:
6377 * pkgconfig/gstreamer-cdda-uninstalled.pc.in:
6378 * pkgconfig/gstreamer-fft-uninstalled.pc.in:
6379 * pkgconfig/gstreamer-floatcast-uninstalled.pc.in:
6380 * pkgconfig/gstreamer-interfaces-uninstalled.pc.in:
6381 * pkgconfig/gstreamer-netbuffer-uninstalled.pc.in:
6382 * pkgconfig/gstreamer-pbutils-uninstalled.pc.in:
6383 * pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
6384 * pkgconfig/gstreamer-riff-uninstalled.pc.in:
6385 * pkgconfig/gstreamer-rtp-uninstalled.pc.in:
6386 * pkgconfig/gstreamer-rtsp-uninstalled.pc.in:
6387 * pkgconfig/gstreamer-sdp-uninstalled.pc.in:
6388 * pkgconfig/gstreamer-tag-uninstalled.pc.in:
6389 * pkgconfig/gstreamer-video-uninstalled.pc.in:
6390 Add srcdir to includes for out-of-source builds
6391 When you use gstreamer uninstalled and build outside
6392 the source tree, the includes need to be specified for
6393 both the source tree and the build tree.
6394 Signed-off-by: David Schleef <ds@schleef.org>
6396 2009-02-22 17:23:52 +0000 Jan Schmidt <thaytan@noraisin.net>
6399 * docs/libs/Makefile.am:
6400 * docs/plugins/Makefile.am:
6401 Use shave for the build output
6403 2009-02-23 12:17:07 +0100 Edward Hervey <bilboed@bilboed.com>
6405 * win32/common/libgstrtsp.def:
6406 win32: Add new symbol to libgstrtsp.def
6408 2009-02-23 10:57:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6410 * gst-libs/gst/rtsp/gstrtspextension.c:
6411 * gst-libs/gst/rtsp/gstrtspextension.h:
6412 Add method for handling server requests
6413 Add a receive_request so that extensions can react to server requests.
6415 2009-02-22 19:20:40 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6417 * tests/check/libs/netbuffer.c:
6418 Correctly cast to GstBuffer * before passing to gst_buffer_(copy|unref)
6420 2009-02-22 19:19:04 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6422 * ext/theora/theoraparse.c:
6423 theoraparse: Use the correct unref functions
6425 2009-02-22 19:18:41 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6427 * sys/ximage/ximagesink.c:
6428 * sys/xvimage/xvimagesink.c:
6429 x(v)imagesink: Correctly cast to GstBuffer * before passing to gst_buffer_unref()
6431 2009-02-22 19:12:00 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6433 * gst-libs/gst/tag/gsttagdemux.c:
6434 tagdemux: Unref the actual buffer instead of the memory address of the buffer
6436 2009-02-22 15:47:53 +0000 Jan Schmidt <thaytan@noraisin.net>
6439 Automatic update of common submodule
6440 From 5d7c9cc to 9cf8c9b
6442 2009-02-22 14:49:29 +0100 Edward Hervey <bilboed@bilboed.com>
6444 * win32/common/libgstrtsp.def:
6445 * win32/common/libgstvideo.def:
6446 win32/common: Update .def files for recent API addition
6448 2009-02-22 13:43:35 +0100 Edward Hervey <bilboed@bilboed.com>
6450 * tests/check/libs/rtp.c:
6451 tests: Fix indentation
6453 2009-02-22 13:42:33 +0100 Edward Hervey <bilboed@bilboed.com>
6455 * gst-libs/gst/video/video.c:
6456 libs/video: Fix gst_video_format_new_caps* functions.
6457 Only add a 'interlaced=True' property to caps *IF* it is interlaced, else
6460 2009-02-21 11:13:36 -0800 David Schleef <ds@schleef.org>
6463 Automatic update of common submodule
6464 From 80c627d to 5d7c9cc
6466 2009-02-20 17:26:40 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6468 * gst-libs/gst/rtsp/gstrtspmessage.c:
6469 Improve key/value parsing
6470 Improve header field parsing by keeping a ref to the key/value instead of
6471 copying it into a local variable.
6473 2009-02-20 12:35:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6475 * gst-libs/gst/rtsp/gstrtspconnection.c:
6476 Add trailing \0 to message length
6477 We always put a trailing 0 at the end of the message body. Reflect this fact in
6478 the length of the message.
6480 2009-02-20 09:50:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6482 * gst-libs/gst/rtsp/gstrtspconnection.c:
6483 Don't parse headers for data messages
6484 Don't try to parse the headers on a data message because they don't have
6487 2009-02-19 12:18:29 -0800 Benjamin M. Schwartz <bens@alum.mit.edu>
6489 * ext/theora/gsttheoraenc.h:
6490 * ext/theora/theoraenc.c:
6491 theoraenc: Add property for speed level control
6492 Add property "speed-level" to control the amount of motion searching
6493 the encoder does. This is only available in libtheora >= 1.0 and
6494 will silently fail with earlier libraries. Fixes: #572275.
6495 Signed-off-by: David Schleef <ds@schleef.org>
6497 2009-02-19 17:40:45 +0100 Edward Hervey <bilboed@bilboed.com>
6499 * gst-libs/gst/video/video.c:
6500 * gst-libs/gst/video/video.h:
6501 video: Fix 'Since' tags
6503 2009-01-26 10:30:53 +0100 Edward Hervey <bilboed@bilboed.com>
6505 * docs/libs/gst-plugins-base-libs-sections.txt:
6506 * gst-libs/gst/video/video.c:
6507 * gst-libs/gst/video/video.h:
6508 video: Add flags for interlaced video along with convenience methods for interlaced caps.
6509 These three flags allow all know combinations of interlaced formats. They should
6510 only be used when the caps contain 'interlaced=True'.
6511 Fixes #163577 (yes, it's a 4 year old bug).
6513 2009-02-19 15:51:52 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6515 * docs/libs/gst-plugins-base-libs-sections.txt:
6516 * gst-libs/gst/rtsp/gstrtspconnection.c:
6517 * gst-libs/gst/rtsp/gstrtspconnection.h:
6518 Make RTSPConnection opaque and rename RTSPChannel
6519 Make the RTSPConnection object opaque so that we can extend it in the future.
6520 Rename GstRTSPChannel to GstRTSPWatch to avoid confusing with the RTSP channels.
6522 2009-01-26 10:31:14 +0100 Edward Hervey <bilboed@bilboed.com>
6524 * gst-libs/gst/riff/riff-media.c:
6525 Add some more mappings for h264 in riff
6527 2009-02-19 10:49:56 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6529 * win32/common/libgstrtsp.def:
6530 Add new RTSP symbols to def files
6531 Add the new RTSP symbols to the windows def file.
6533 2009-02-19 10:44:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6535 * docs/libs/gst-plugins-base-libs-sections.txt:
6536 * gst-libs/gst/app/gstappsink.c:
6537 * gst-libs/gst/app/gstappsink.h:
6538 * tests/check/Makefile.am:
6539 * tests/check/elements/.gitignore:
6540 * tests/check/elements/appsink.c:
6541 Add method to install callbacks on appsink
6542 Based on pacth by Martin Samuelsson <martin dot samuelsson at axis dot com>
6544 Add gst_app_sink_set_callbacks() to install a set of callbacks. This is a more
6545 performant alternative to connecting to the signals.
6546 Add a unit test for appsink.
6547 Clean up some of the appsink docs.
6548 API: GstAppSink::gst_app_sink_set_callbacks()
6550 2009-02-18 18:46:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6552 * docs/libs/gst-plugins-base-libs-sections.txt:
6553 * gst-libs/gst/rtsp/gstrtspconnection.c:
6554 * gst-libs/gst/rtsp/gstrtspconnection.h:
6555 Add RTSP accept method
6556 Add a method to accept a connection on a socket and create a GstRTSPConnection
6558 API: gst_rtsp_connection_accept()
6560 2009-02-18 17:42:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6562 * docs/libs/gst-plugins-base-libs-sections.txt:
6563 * gst-libs/gst/rtsp/gstrtspconnection.c:
6564 * gst-libs/gst/rtsp/gstrtspconnection.h:
6565 Add RTSP channel object for async io
6566 Add a GstRTSPChannel object that wraps a GSource around the RTSP connection so
6567 that the connection can be monitored from a maincontext. This allows us to
6568 operate in ASYNC mode, which is handy when building a server.
6569 Rework the old code to use the async code under the hood.
6570 API: gst_rtsp_channel_new()
6571 API: gst_rtsp_channel_unref()
6572 API: gst_rtsp_channel_attach()
6573 API: gst_rtsp_channel_queue_message()
6575 2009-02-15 07:30:17 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6577 * gst/audioresample/gstaudioresample.c:
6578 audioresample: Add locking to protect the resampling context
6579 When setting the quality/filter-length while PLAYING the
6580 resampling context will be destroyed and created again in
6581 some cases, which will cause crashes in the transform function
6582 if it's called at that time.
6584 2009-02-13 10:10:25 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6586 * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
6587 * gst/videotestsrc/videotestsrc.c:
6588 ffmpegcolorspace/videotestsrc: Use v308 instead of V308
6590 2009-02-12 19:02:59 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6592 * gst/ffmpegcolorspace/avcodec.h:
6593 * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
6594 * gst/ffmpegcolorspace/imgconvert.c:
6595 * gst/ffmpegcolorspace/imgconvert_template.h:
6596 ffmpegcolorspace: Add support for packed 4:4:4 YUV (format=V308)
6597 Only conversions from/to are implemented, which
6598 gives (indirect) support for all possible conversions.
6599 Partially fixes bug #571147.
6601 2009-02-12 18:17:53 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6603 * gst/videotestsrc/videotestsrc.c:
6604 videotestsrc: Add support for packed 4:4:4 YUV (format=V308)
6605 Partially fixes bug #571147.
6607 2009-02-12 09:18:20 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6609 * gst-libs/gst/tag/gsttagdemux.c:
6610 tagdemux: don't abort when downstream pulls a buffer of size 0
6611 Pulling a 0-sized buffer is allowed, and we should handle this correctly instead of
6612 aborting. Fixes #571009 (wma file with ID3v2 tag).
6614 2009-02-11 16:39:55 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6616 * gst-libs/gst/riff/riff-read.c:
6617 riff: error out on nonsensical chunk sizes instead of aborting
6618 When encountering a nonsensical chunk size such as (guint)-1, error out cleanly instead of
6619 continuing and trying to g_memdup() 4GB of data that doesn't exist, which will either abort
6620 in g_malloc() or crash.
6621 Fixes #553295, crash with fuzzed AVI file.
6623 2009-02-11 16:39:06 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6626 Make git ignore backup files.
6628 2009-02-10 20:38:58 -0800 Michael Smith <msmith@syncword.(none)>
6630 * gst/playback/gstplaybin2.c:
6631 Revert "Remove pad-removed handlers after setting the decodebins to NULL."
6632 This reverts commit b36d8f3e119f9edc5993c08025614ee32642972e.
6633 This brought back some deadlocks. A small leak is better, for now. Need to
6634 figure out a way to fix the leak properly.
6636 2009-02-10 17:16:07 -0800 Michael Smith <msmith@songbirdnest.com>
6638 * gst/playback/gstplaybin2.c:
6639 playbin2: Fix segfault on notify after group change.
6640 If our group has been switched, then we get a selector active-pad
6641 notification, we don't need to notify.
6643 2009-02-10 17:10:33 -0800 Michael Smith <msmith@songbirdnest.com>
6645 * gst/playback/gstplaysink.c:
6646 playbin2: Look for volume/mute properties recursively in audio element.
6647 Rather than only checking for volume property on the audio sink
6648 directly, recursively look for it on sinks within it (if it's a bin).
6649 Allows use of sink-as-volume-control where the application has supplied
6650 an audio-sink bin that includes a real audio sink internally.
6652 2009-02-10 18:29:22 +0000 Christian Schaller <cschalle@crazyhorse.localdomain>
6654 * gst-plugins-base.spec.in:
6655 Update spec file with latest additions and changes, most noteably the move of appsrc appsink into -base
6657 2009-02-10 17:39:45 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6659 * gst/videotestsrc/videotestsrc.c:
6660 videotestsrc: Add support for Y444 (planar 4:4:4 YUV)
6661 Partially fixes bug #571147.
6663 2009-02-10 17:37:06 +0100 Peter Kjellerstedt <pkj@axis.com>
6665 * gst-libs/gst/rtsp/gstrtspmessage.c:
6666 gstrtspmessage: Minor documentation correction.
6667 Corrected documentation about what needs to be freed after calling
6668 gst_rtsp_message_new(), gst_rtsp_message_new_request(),
6669 gst_rtsp_message_new_response() and gst_rtsp_message_new_data().
6671 2009-02-10 11:00:12 +0100 Antoine Tremblay <hexa00@gmail.com>
6673 * ext/alsa/gstalsamixer.c:
6674 alsamixer: Fix race condition that made alsamixer not working properly
6675 This is due to race conditions between functions that
6676 modified the mixer like set_volume and
6677 snd_mixer_handle_events since the handle_events
6678 can now be called at any time.
6679 Fixed by adding locking around any snd_mixer call
6680 since even read functions can modify the mixer stucture, since
6681 alsa likes to clear it's values before reading new ones.
6682 The favorite race condition seemed to be that set_volume
6683 called read_elem (in alsalib) that reset the volumes to
6684 0 and then read them with read_x_volume. This read looped
6685 on each channel and as the race condition occured the
6686 channels value could be anything , most of the time
6687 it was 0. Thus no value was read or only the value of
6688 one channel was and the volume was reset to 0.
6691 2009-02-09 12:02:21 +0100 Edward Hervey <bilboed@bilboed.com>
6694 Bump revision to use for common submodule.
6696 2009-02-05 15:47:00 +0200 Stefan Kost <ensonic@users.sf.net>
6698 * sys/xvimage/xvimagesink.c:
6699 xvimagesink: do not call _xwindow_clear on ready->paused.
6700 Calling clear at that transition does things like stopping xvideo (which is not
6701 running at that time) and also clearing anything what the application might have drawn.
6702 This breaks handle-expose and autopaint-colorkey features.
6704 2009-02-04 17:03:52 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6706 * docs/libs/gst-plugins-base-libs-sections.txt:
6707 * gst-libs/gst/rtsp/gstrtsprange.c:
6708 * gst-libs/gst/rtsp/gstrtsprange.h:
6709 RTSPRange: Add method to serialize ranges
6710 Add gst_rtsp_range_to_string() to serialize a GstRTSPRange to a string that can
6711 be used by a server.
6712 API: GstRTSPRange::gst_rtsp_range_to_string()
6714 2009-02-04 13:16:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6716 * gst-libs/gst/rtsp/gstrtspurl.c:
6717 * gst-libs/gst/rtsp/gstrtspurl.h:
6718 GstRTSPUrl: Add some const to methods
6719 Add const to the methods that do not modify the object.
6721 2009-02-04 13:53:30 +0200 Stefan Kost <ensonic@users.sf.net>
6723 * gst/playback/gstplaysink.c:
6724 playbin2: implement GST_PLAY_FLAG_NATIVE_{AUDIO,VIDEO}
6725 The flags where present but actually not been taken into account.
6727 2009-02-04 12:06:38 +0200 Stefan Kost <ensonic@users.sf.net>
6729 * gst/audioresample/gstaudioresample.c:
6730 audioresample: Add a proper deprecation comment and also drop G_PARAM_CONSTRUCT.
6731 The comment will ensure that is is marked properly in the docs and the
6732 GParamSpecflag was causing a duplicated initialisation of the same value.
6734 2009-02-04 11:18:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6736 * gst-libs/gst/rtsp/gstrtspconnection.c:
6737 Add more g_return_if_fail() calls
6738 Check that we have a valid file descriptor before entering certain functions in
6739 order to avoid undesirable situations.
6740 Add some more debugging in the connect method.
6742 2009-02-04 10:31:21 +0200 Stefan Kost <ensonic@users.sf.net>
6745 * gst/audioresample/Makefile.am:
6746 * gst/audioresample/gstaudioresample.c:
6747 audioresample: Only pull in liboil if its actualy used.
6748 Liboil still has quite significant startup overhead especialy on embedded
6749 platforms. In audioresample it was only used for the profiling timer.
6751 2009-02-03 15:26:08 +0200 Stefan Kost <ensonic@users.sf.net>
6753 * gst/typefind/gsttypefindfunctions.c:
6754 typefind: Make the flac check more tight to not mistace some aac files for flac. Fixes #570356.
6755 Add comments about the flac format. Tighten the check to not allow values that
6758 2009-02-03 10:52:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6760 * win32/common/libgstrtsp.def:
6762 Add new methods to the windows def file.
6764 2009-02-02 17:25:21 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
6766 * gst-libs/gst/pbutils/install-plugins.c:
6767 * tests/check/libs/pbutils.c:
6768 pbutils: remove duplicate detail strings when calling the external codec installer
6769 It doesn't make sense to ask installers for the same codec or element twice, so filter out duplicate requests before calling the external helper script and make the unit test check this works right. Fixes #567636.
6771 2009-02-02 18:05:42 +0200 Stefan Kost <ensonic@users.sf.net>
6773 * gst-libs/gst/audio/gstaudiosink.c:
6774 * gst-libs/gst/audio/gstaudiosink.h:
6775 Add a FIXME 0.11. Make the log message a bit more detailed and add comments.
6777 2009-02-02 15:43:03 +0200 Stefan Kost <ensonic@users.sf.net>
6780 * gst/audioresample/gstaudioresample.c:
6781 Allow to configure the resampler function for integer to skip the benchmarking. Fix releasing the intger resampler in benchmark.
6783 2009-02-02 13:30:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6785 * sys/ximage/ximagesink.c:
6786 Fix buffer_alloc in ximagesink
6787 Remove some useless debug info that reported wrong image sizes.
6788 When upstream does not accept out suggested size, fall back to allocating an
6789 image of the requested width/height instead of the currently configured size.
6790 The problem is that an image is reused from the pool because the width/height
6791 match but the caps on the new buffer are the requested caps with possibly
6792 different height/width resulting in errors.
6794 2009-02-02 12:54:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6796 * gst/playback/gstdecodebin2.c:
6797 * gst/playback/gsturidecodebin.c:
6798 Fix documentation for autoplug-select
6799 fix the documentation strings for the autoplug-select signal.
6802 2009-02-02 10:09:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6804 * gst-libs/gst/rtsp/gstrtspmessage.c:
6805 Fix string leak in rtspmessage
6806 when we remove a header field from a message we must free the value associated
6807 with the key to avoid a memory leak.
6809 2009-01-31 18:45:47 +0200 Stefan Kost <ensonic@users.sf.net>
6811 * docs/libs/gst-plugins-base-libs-docs.sgml:
6812 Its "Base Library" and not just "Library".
6814 2009-01-31 18:44:32 +0200 Stefan Kost <ensonic@users.sf.net>
6816 * gst-libs/gst/audio/gstaudiofilter.c:
6817 Link to the class, as we can't link to the members yet.
6819 2009-01-30 17:48:23 -0800 Michael Smith <msmith@songbirdnest.com>
6821 * gst/playback/gstplaybin2.c:
6822 Remove pad-removed handlers after setting the decodebins to NULL.
6823 They do needed cleanup; without this we leak selector requestpads.
6825 2009-01-30 17:47:07 -0800 Michael Smith <msmith@songbirdnest.com>
6827 * gst/playback/gstplaybin2.c:
6828 Unref selector request pad even if we no longer have a selector.
6829 During destruction, we won't have a selector any more, but we still need
6830 to unref the pad to avoid leaking it.
6832 2009-01-30 15:23:23 -0800 Michael Smith <msmith@songbirdnest.com>
6834 * gst/playback/gstplaybin2.c:
6835 Unref source in playbin2's finalize method
6837 2009-01-30 12:04:01 -0800 Michael Smith <msmith@songbirdnest.com>
6839 * gst/playback/gstplaysink.c:
6840 Fix more leaks of pads and elements in gstplaysink.
6841 Don't keep extra references to volume and mute elements; we don't need
6843 Ensure we unref pads that we have references to, and release request
6846 2009-01-30 11:04:37 -0800 Michael Smith <msmith@songbirdnest.com>
6848 * gst/playback/gstplaysink.c:
6849 Avoid leaking all playsinks. Fix some internal leaks.
6850 Playsink was holding references to itself. Don't do that, it's not cool.
6851 Also, free all chains in dispose.
6853 2009-01-30 10:54:12 -0800 Michael Smith <msmith@songbirdnest.com>
6855 * gst/playback/gstplaybin2.c:
6856 Unref peer request pad after releasing it, since we hold a reference.
6858 2009-01-30 10:52:52 -0800 Michael Smith <msmith@songbirdnest.com>
6860 * gst/playback/gstplaybin2.c:
6861 Fix caps leak in playbin2.
6863 2009-01-30 10:51:11 -0800 Michael Smith <msmith@songbirdnest.com>
6865 * gst/playback/gstplaybin2.c:
6866 Unref active pad from selector when finding active stream.
6868 2009-01-30 10:49:55 -0800 Michael Smith <msmith@songbirdnest.com>
6870 * gst/playback/gstplaybin2.c:
6871 Free uris when finalizing playbin2 instance.
6873 2009-01-30 10:38:17 -0800 Michael Smith <msmith@songbirdnest.com>
6875 * gst/playback/gsturidecodebin.c:
6876 Unref pads when iterating over them in analyse_source.
6877 Fixes leak of source's srcpad when using uridecodebin.
6879 2009-01-30 22:22:07 +0200 Stefan Kost <ensonic@users.sf.net>
6881 * docs/plugins/gst-plugins-base-plugins-docs.sgml:
6882 Add releaseinfo with online url.
6884 2009-01-30 17:58:15 +0000 Jan Schmidt <jan.schmidt@sun.com>
6886 * gst/playback/gstplaybasebin.c:
6887 Fix compilation warning on Forte
6889 2009-01-30 17:16:39 +0000 Jan Schmidt <jan.schmidt@sun.com>
6891 * gst/adder/gstadder.c:
6892 Don't do void pointer arithmetic.
6894 2009-01-30 17:25:51 +0000 Jan Schmidt <thaytan@noraisin.net>
6899 2009-01-30 08:50:53 +0100 Edward Hervey <bilboed@bilboed.com>
6903 Use a symbolic link for the pre-commit client-side hook
6905 2009-01-30 08:12:42 +0100 Edward Hervey <bilboed@bilboed.com>
6908 Add more files/directories to ignore
6910 2009-01-29 14:00:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6912 * gst-libs/gst/rtsp/gstrtspdefs.c:
6914 Fix some typos in the doc string of the new
6915 gst_rtsp_options_as_string() method.
6917 2009-01-29 11:55:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6919 * docs/libs/gst-plugins-base-libs-sections.txt:
6920 * gst-libs/gst/rtsp/gstrtspconnection.c:
6921 * gst-libs/gst/rtsp/gstrtspmessage.c:
6922 * gst-libs/gst/rtsp/gstrtspmessage.h:
6923 Add new RTSP message method to set header
6924 Add gst_rtsp_message_take_header() that takes ownership of the passed header
6925 value. This allows us to avoid an allocations and memory copy in some
6927 API: GstRTSPMessage::gst_rtsp_message_take_header()
6929 2009-01-29 11:51:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6931 * docs/libs/gst-plugins-base-libs-sections.txt:
6932 Add new method to docs
6933 Add the new gst_rtsp_options_as_text() method to the docs.
6935 2009-01-28 11:48:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
6937 * gst-libs/gst/rtsp/gstrtspdefs.c:
6938 * gst-libs/gst/rtsp/gstrtspdefs.h:
6939 Add method to serialize RTSP options
6940 Add gst_rtsp_options_as_text() method to serialize a set of RTSP options to a
6942 API: GstRTSP::gst_rtsp_options_as_text()
6944 2009-01-26 17:59:37 -0800 Michael Smith <msmith@songbirdnest.com>
6946 * gst/typefind/gsttypefindfunctions.c:
6947 Ensure we have sufficient data when using data scan contexts.
6948 Fixes crashes typefinding things that look like they might contain AAC
6949 data (but probably aren't actually AAC).
6951 2009-01-26 23:32:09 +0000 Jan Schmidt <thaytan@noraisin.net>
6953 * ext/gio/Makefile.am:
6954 Fix include order for gio plugin
6956 2009-01-23 23:59:48 +0000 Jan Schmidt <thaytan@noraisin.net>
6958 * win32/common/config.h:
6959 Update win32 config.h for 0.10.22.1 dev cycle
6961 2009-01-23 23:16:11 +0000 Jan Schmidt <thaytan@noraisin.net>
6964 * docs/libs/.gitignore:
6965 * gst-libs/gst/audio/.gitignore:
6966 * gst-libs/gst/video/.gitignore:
6968 * tests/examples/dynamic/.gitignore:
6969 Extend and clean up git ignores
6971 2009-01-23 12:31:06 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
6974 * docs/plugins/Makefile.am:
6975 * docs/plugins/gst-plugins-base-plugins-sections.txt:
6976 * docs/plugins/gst-plugins-base-plugins.args:
6977 * docs/plugins/gst-plugins-base-plugins.hierarchy:
6978 * docs/plugins/gst-plugins-base-plugins.interfaces:
6979 * docs/plugins/gst-plugins-base-plugins.prerequisites:
6980 * docs/plugins/inspect/plugin-adder.xml:
6981 * docs/plugins/inspect/plugin-alsa.xml:
6982 * docs/plugins/inspect/plugin-app.xml:
6983 * docs/plugins/inspect/plugin-audioconvert.xml:
6984 * docs/plugins/inspect/plugin-audiorate.xml:
6985 * docs/plugins/inspect/plugin-audioresample.xml:
6986 * docs/plugins/inspect/plugin-audiotestsrc.xml:
6987 * docs/plugins/inspect/plugin-cdparanoia.xml:
6988 * docs/plugins/inspect/plugin-decodebin.xml:
6989 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
6990 * docs/plugins/inspect/plugin-gdp.xml:
6991 * docs/plugins/inspect/plugin-gio.xml:
6992 * docs/plugins/inspect/plugin-gnomevfs.xml:
6993 * docs/plugins/inspect/plugin-libvisual.xml:
6994 * docs/plugins/inspect/plugin-ogg.xml:
6995 * docs/plugins/inspect/plugin-pango.xml:
6996 * docs/plugins/inspect/plugin-playback.xml:
6997 * docs/plugins/inspect/plugin-queue2.xml:
6998 * docs/plugins/inspect/plugin-subparse.xml:
6999 * docs/plugins/inspect/plugin-tcp.xml:
7000 * docs/plugins/inspect/plugin-theora.xml:
7001 * docs/plugins/inspect/plugin-typefindfunctions.xml:
7002 * docs/plugins/inspect/plugin-uridecodebin.xml:
7003 * docs/plugins/inspect/plugin-video4linux.xml:
7004 * docs/plugins/inspect/plugin-videorate.xml:
7005 * docs/plugins/inspect/plugin-videoscale.xml:
7006 * docs/plugins/inspect/plugin-videotestsrc.xml:
7007 * docs/plugins/inspect/plugin-volume.xml:
7008 * docs/plugins/inspect/plugin-vorbis.xml:
7009 * docs/plugins/inspect/plugin-ximagesink.xml:
7010 * docs/plugins/inspect/plugin-xvimagesink.xml:
7011 * gst/audioresample/Makefile.am:
7012 * gst/audioresample/README:
7013 * gst/audioresample/arch.h:
7014 * gst/audioresample/buffer.c:
7015 * gst/audioresample/buffer.h:
7016 * gst/audioresample/debug.c:
7017 * gst/audioresample/debug.h:
7018 * gst/audioresample/fixed_arm4.h:
7019 * gst/audioresample/fixed_arm5e.h:
7020 * gst/audioresample/fixed_bfin.h:
7021 * gst/audioresample/fixed_debug.h:
7022 * gst/audioresample/fixed_generic.h:
7023 * gst/audioresample/functable.c:
7024 * gst/audioresample/functable.h:
7025 * gst/audioresample/gstaudioresample.c:
7026 * gst/audioresample/gstaudioresample.h:
7027 * gst/audioresample/resample.c:
7028 * gst/audioresample/resample.h:
7029 * gst/audioresample/resample_chunk.c:
7030 * gst/audioresample/resample_functable.c:
7031 * gst/audioresample/resample_ref.c:
7032 * gst/audioresample/resample_sse.h:
7033 * gst/audioresample/speex_resampler.h:
7034 * gst/audioresample/speex_resampler_double.c:
7035 * gst/audioresample/speex_resampler_float.c:
7036 * gst/audioresample/speex_resampler_int.c:
7037 * gst/audioresample/speex_resampler_wrapper.h:
7038 * gst/speexresample/Makefile.am:
7039 * gst/speexresample/README:
7040 * gst/speexresample/arch.h:
7041 * gst/speexresample/fixed_arm4.h:
7042 * gst/speexresample/fixed_arm5e.h:
7043 * gst/speexresample/fixed_bfin.h:
7044 * gst/speexresample/fixed_debug.h:
7045 * gst/speexresample/fixed_generic.h:
7046 * gst/speexresample/gstspeexresample.c:
7047 * gst/speexresample/gstspeexresample.h:
7048 * gst/speexresample/resample.c:
7049 * gst/speexresample/resample_sse.h:
7050 * gst/speexresample/speex_resampler.h:
7051 * gst/speexresample/speex_resampler_double.c:
7052 * gst/speexresample/speex_resampler_float.c:
7053 * gst/speexresample/speex_resampler_int.c:
7054 * gst/speexresample/speex_resampler_wrapper.h:
7055 * gst/typefind/gsttypefindfunctions.c:
7056 * tests/check/Makefile.am:
7057 * tests/check/elements/audioresample.c:
7058 * tests/check/elements/speexresample.c:
7059 Rename files and types from speexresample to audioresample
7060 Rename files and types from speexresample to audioresample
7061 to finish the move and to prevent any confusion.
7063 2009-01-23 11:44:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7065 * sys/xvimage/xvimagesink.c:
7066 Add some more debugging to the Xv strides
7067 Add some more debugging to the strides as they are received from the server and
7068 the expected strides.
7070 2009-01-23 11:40:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7072 * gst/typefind/gsttypefindfunctions.c:
7073 Add typefind function for gsm
7074 Because core now supports typefindfactories without a typefind function we can
7075 register a factory fo GSM that will --if all else fails-- assume the file is a
7076 GSM file based on the registered extension.
7079 2009-01-23 11:37:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7081 * gst/playback/gsturidecodebin.c:
7082 Use more performant link function
7083 We can use gst_element_link_pads() instead of the more generic
7084 gst_element_link() function because we know the pads. This saves some cycles
7085 because the more generic function needs to search for possible compatible caps
7088 2009-01-23 11:33:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7090 * gst-libs/gst/riff/riff-ids.h:
7091 * gst-libs/gst/riff/riff-media.c:
7092 Add more codec ids for RIFF formats
7093 Handle codec ID for various other AAC formats.
7094 Sync the list of possible codec ids with that of ffmpeg.
7097 2009-01-23 11:27:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7099 * ext/theora/theoradec.c:
7100 Use rounded values for image strides and sizes
7101 Round up the height before calculating the expected size and
7102 strides of the output image.
7104 2009-01-23 11:23:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7106 * ext/alsa/gstalsasink.c:
7107 Improve debug message
7108 Improve the debug message when alsa returns an error.
7110 2009-01-23 11:07:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7112 * gst-libs/gst/app/gstappsrc.c:
7113 Reset queued_bytes counter when flushing
7114 Set the amount of queued bytes in the internal queue back to 0 when we clear the
7118 2009-01-23 10:19:27 +0100 Benjamin Gaignard <benjamin@gaignard.net>
7120 * gst/typefind/gsttypefindfunctions.c:
7121 Add typefinder for Mobile XMF. Fixes bug #568707.
7123 2009-01-23 10:00:11 +0100 Brian Cameron <brian.cameron@sun.com>
7126 Fix linking on Solaris. Fixes bug #568482.
7127 Check for nsl and socket libraries and add them to
7128 LIBS if they're found. They're needed for socket()
7129 and gethostbyname() on Solaris.
7131 2009-01-22 22:09:47 +0000 Jan Schmidt <thaytan@noraisin.net>
7133 * gst/playback/gstplaybasebin.c:
7134 Fix use-after-unref problem noticed by Josep Torra Valles, and run
7137 2009-01-22 17:46:59 +0200 Stefan Kost <ensonic@users.sf.net>
7140 Update common snapshot.
7142 2009-01-22 13:47:24 +0100 Sebastian Dröge <slomo@circular-chaos.org>
7147 2009-01-22 13:12:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7149 Merge branch 'master' of ssh://git.freedesktop.org/git/gstreamer/gst-plugins-base
7151 2009-01-22 10:14:28 +0100 Sebastian Dröge <slomo@circular-chaos.org>
7153 * gst-libs/gst/fft/gstfftf32.c:
7154 * gst-libs/gst/fft/gstfftf64.c:
7155 * gst-libs/gst/fft/gstffts16.c:
7156 * gst-libs/gst/fft/gstffts32.c:
7157 Reduce the number of allocations for creating FFT contexts
7158 Reduce the number of allocations from 2 to 1 for every FFT
7159 context by allocating enough memory for the FFT context
7160 and passing parts of it to the kissfft allocation functions.
7162 2009-01-22 11:32:56 +0000 Jan Schmidt <thaytan@noraisin.net>
7165 Back to devel -> 0.10.22.1
7167 2009-01-22 05:57:53 +0100 Edward Hervey <bilboed@bilboed.com>
7171 Install and use pre-commit indentation hook from common
7173 2009-01-21 13:09:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
7175 * gst-libs/gst/rtp/gstrtpbuffer.c:
7176 * tests/check/libs/rtp.c:
7177 Avoid overflows in the padding checks by doing the check slightly
7179 Add a unit test to check for correct behaviour.
7181 2009-01-21 04:31:32 +0100 Edward Hervey <bilboed@bilboed.com>
7184 autogen.sh : Use git submodule
7186 === release 0.10.22 ===
7188 2009-01-19 23:10:50 +0000 Jan Schmidt <thaytan@mad.scientist.com>
7194 * docs/plugins/gst-plugins-base-plugins.hierarchy:
7195 * docs/plugins/gst-plugins-base-plugins.interfaces:
7196 * docs/plugins/gst-plugins-base-plugins.prerequisites:
7197 * docs/plugins/inspect/plugin-adder.xml:
7198 * docs/plugins/inspect/plugin-alsa.xml:
7199 * docs/plugins/inspect/plugin-app.xml:
7200 * docs/plugins/inspect/plugin-audioconvert.xml:
7201 * docs/plugins/inspect/plugin-audiorate.xml:
7202 * docs/plugins/inspect/plugin-audioresample.xml:
7203 * docs/plugins/inspect/plugin-audiotestsrc.xml:
7204 * docs/plugins/inspect/plugin-cdparanoia.xml:
7205 * docs/plugins/inspect/plugin-decodebin.xml:
7206 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
7207 * docs/plugins/inspect/plugin-gdp.xml:
7208 * docs/plugins/inspect/plugin-gnomevfs.xml:
7209 * docs/plugins/inspect/plugin-libvisual.xml:
7210 * docs/plugins/inspect/plugin-ogg.xml:
7211 * docs/plugins/inspect/plugin-pango.xml:
7212 * docs/plugins/inspect/plugin-playback.xml:
7213 * docs/plugins/inspect/plugin-queue2.xml:
7214 * docs/plugins/inspect/plugin-subparse.xml:
7215 * docs/plugins/inspect/plugin-tcp.xml:
7216 * docs/plugins/inspect/plugin-theora.xml:
7217 * docs/plugins/inspect/plugin-typefindfunctions.xml:
7218 * docs/plugins/inspect/plugin-uridecodebin.xml:
7219 * docs/plugins/inspect/plugin-video4linux.xml:
7220 * docs/plugins/inspect/plugin-videorate.xml:
7221 * docs/plugins/inspect/plugin-videoscale.xml:
7222 * docs/plugins/inspect/plugin-videotestsrc.xml:
7223 * docs/plugins/inspect/plugin-volume.xml:
7224 * docs/plugins/inspect/plugin-vorbis.xml:
7225 * docs/plugins/inspect/plugin-ximagesink.xml:
7226 * docs/plugins/inspect/plugin-xvimagesink.xml:
7227 * gst-plugins-base.doap:
7257 * win32/common/config.h:
7259 Original commit message from CVS:
7262 2009-01-19 22:01:01 +0000 Jan Schmidt <thaytan@mad.scientist.com>
7294 Original commit message from CVS:
7297 2009-01-16 11:44:04 +0000 Sebastian Dröge <slomo@circular-chaos.org>
7299 gst-libs/gst/fft/: Use correct struct alignment everywhere to prevent unaligned memory accesses, resulting in SIGBUS ...
7300 Original commit message from CVS:
7301 * gst-libs/gst/fft/_kiss_fft_guts_f32.h:
7302 * gst-libs/gst/fft/_kiss_fft_guts_f64.h:
7303 * gst-libs/gst/fft/_kiss_fft_guts_s16.h:
7304 * gst-libs/gst/fft/_kiss_fft_guts_s32.h:
7305 * gst-libs/gst/fft/kiss_fftr_f32.c: (kiss_fftr_f32_alloc):
7306 * gst-libs/gst/fft/kiss_fftr_f64.c: (kiss_fftr_f64_alloc):
7307 * gst-libs/gst/fft/kiss_fftr_s16.c: (kiss_fftr_s16_alloc):
7308 * gst-libs/gst/fft/kiss_fftr_s32.c: (kiss_fftr_s32_alloc):
7309 Use correct struct alignment everywhere to prevent unaligned
7310 memory accesses, resulting in SIGBUS on sparc and probably others.
7313 2009-01-16 11:40:02 +0000 Sebastian Dröge <slomo@circular-chaos.org>
7315 gst-libs/gst/tag/gsttagdemux.c: Forward unknown events upstream to allow latency configuration.
7316 Original commit message from CVS:
7317 * gst-libs/gst/tag/gsttagdemux.c: (gst_tag_demux_srcpad_event):
7318 Forward unknown events upstream to allow latency configuration.
7321 2009-01-13 14:47:19 +0000 Wim Taymans <wim.taymans@gmail.com>
7323 gst/playback/gstplaybin2.c: Provide the right arguments to a debug line.
7324 Original commit message from CVS:
7325 * gst/playback/gstplaybin2.c: (groups_set_locked_state):
7326 Provide the right arguments to a debug line.
7328 2009-01-13 06:51:54 +0000 Stefan Kost <ensonic@users.sourceforge.net>
7330 sys/xvimage/xvimagesink.c: Don't reset the colorkey when element is reused. Fixes #567511.
7331 Original commit message from CVS:
7332 * sys/xvimage/xvimagesink.c:
7333 Don't reset the colorkey when element is reused. Fixes #567511.
7335 2009-01-09 23:42:22 +0000 Jan Schmidt <thaytan@mad.scientist.com>
7337 configure.ac: 0.10.21.3 pre-release
7338 Original commit message from CVS:
7340 0.10.21.3 pre-release
7342 2009-01-09 23:13:17 +0000 Jan Schmidt <thaytan@mad.scientist.com>
7344 gst-libs/gst/app/gstappsink.c: Store the returned signal id in the right slot when registering the pull-buffer signal.
7345 Original commit message from CVS:
7346 * gst-libs/gst/app/gstappsink.c:
7347 Store the returned signal id in the right slot when
7348 registering the pull-buffer signal.
7350 Spotted by: Thomas Vander Stichele <thomas at apestaart dot org>
7352 2009-01-09 17:17:50 +0000 Tim-Philipp Müller <tim@centricular.net>
7354 gst-libs/gst/interfaces/mixer.c: Small docs addition to clarify that one really mustn't free the constant GList retur...
7355 Original commit message from CVS:
7356 * gst-libs/gst/interfaces/mixer.c:
7357 Small docs addition to clarify that one really mustn't free
7358 the constant GList returned (#566812).
7360 2009-01-08 17:18:24 +0000 Wim Taymans <wim.taymans@gmail.com>
7362 Add GType for GstRTSPUrl and expose a copy function because we can.
7363 Original commit message from CVS:
7364 * docs/libs/gst-plugins-base-libs-sections.txt:
7365 * gst-libs/gst/rtsp/gstrtspurl.c: (register_rtsp_url_type),
7366 (gst_rtsp_url_get_type), (gst_rtsp_url_copy):
7367 * gst-libs/gst/rtsp/gstrtspurl.h:
7368 * win32/common/libgstrtsp.def:
7369 Add GType for GstRTSPUrl and expose a copy function because we can.
7370 API: gst_rtsp_url_copy()
7373 2009-01-07 18:36:04 +0000 Sebastian Dröge <slomo@circular-chaos.org>
7375 Add plugin dependency for the GIO and GVfs modules.
7376 Original commit message from CVS:
7378 * ext/gio/gstgio.c: (plugin_init):
7379 Add plugin dependency for the GIO and GVfs modules.
7382 2009-01-07 18:32:33 +0000 Sebastian Dröge <slomo@circular-chaos.org>
7384 Add plugin dependency for the gnomevfs modules.
7385 Original commit message from CVS:
7387 * ext/gnomevfs/gstgnomevfs.c: (plugin_init):
7388 Add plugin dependency for the gnomevfs modules.
7391 2009-01-07 18:30:52 +0000 Sebastian Dröge <slomo@circular-chaos.org>
7393 win32/common/libgstcdda.def: Add new symbol to the list of exported symbols.
7394 Original commit message from CVS:
7395 * win32/common/libgstcdda.def:
7396 Add new symbol to the list of exported symbols.
7398 2009-01-07 13:52:14 +0000 Wim Taymans <wim.taymans@gmail.com>
7400 gst/playback/gstplaybin2.c: Fix some comments and docs.
7401 Original commit message from CVS:
7402 * gst/playback/gstplaybin2.c: (gst_play_bin_class_init),
7403 (gst_play_bin_set_uri), (gst_play_bin_set_suburi),
7404 (no_more_pads_cb), (drained_cb), (group_set_locked_state_unlocked),
7405 (activate_group), (deactivate_group), (groups_set_locked_state),
7406 (gst_play_bin_change_state):
7407 Fix some comments and docs.
7408 Post an error message when we fail to link the selector to the sink.
7409 Remove pushing of EOS, this seems unneeded.
7410 Lock the state of deactivated groups so that they don't accidentally
7411 reactivate when the playbin2 state changes.
7412 Reuse uridecodebins.
7413 Unlock and relock state of groups when playbin goes to NULL.
7416 * gst/playback/gsturidecodebin.c: (pad_removed_cb), (type_found):
7417 Only do something in the pad removed callback when we are dealing with
7418 our sourcepads because the sinkpads don't have a ghostpad.
7420 2009-01-07 10:50:15 +0000 Sebastian Dröge <slomo@circular-chaos.org>
7422 gst-libs/gst/cdda/gstcddabasesrc.*: Make the GType of GstCDDABaseSrcMode public for bindings.
7423 Original commit message from CVS:
7424 * gst-libs/gst/cdda/gstcddabasesrc.c:
7425 * gst-libs/gst/cdda/gstcddabasesrc.h:
7426 Make the GType of GstCDDABaseSrcMode public for bindings.
7429 2009-01-06 18:03:51 +0000 Tim-Philipp Müller <tim@centricular.net>
7431 Use new core API to make registry re-scan the plugin whenever visualisations are added or removed (see #350477).
7432 Original commit message from CVS:
7434 * ext/libvisual/visual.c: (plugin_init):
7435 Use new core API to make registry re-scan the plugin
7436 whenever visualisations are added or removed (see #350477).
7438 2009-01-06 17:30:31 +0000 José Alburquerque <jaalburqu@svn.gnome.org>
7440 gst-libs/gst/audio/gstaudioclock.*: Make gst_audio_clock_new use const gchar* to ease the wrapping of
7441 Original commit message from CVS:
7442 Patch by: José Alburquerque <jaalburqu svn gnome org>
7443 * gst-libs/gst/audio/gstaudioclock.c: (gst_audio_clock_new):
7444 * gst-libs/gst/audio/gstaudioclock.h:
7445 Make gst_audio_clock_new use const gchar* to ease the wrapping of
7446 C++ bindings. Fixes #566723.
7448 2009-01-06 12:16:18 +0000 Sebastian Dröge <slomo@circular-chaos.org>
7450 Add pkg-config files for libgstapp. Fixes bug #566761.
7451 Original commit message from CVS:
7453 * pkgconfig/Makefile.am:
7454 * pkgconfig/gstreamer-app-uninstalled.pc.in:
7455 * pkgconfig/gstreamer-app.pc.in:
7456 Add pkg-config files for libgstapp. Fixes bug #566761.
7458 2009-01-06 11:10:29 +0000 Tim-Philipp Müller <tim@centricular.net>
7460 gst-libs/gst/app/: Make debug categories static. Use _element_class_set_details_simple().
7461 Original commit message from CVS:
7462 * gst-libs/gst/app/gstappsink.c:
7463 * gst-libs/gst/app/gstappsink.h:
7464 * gst-libs/gst/app/gstappsrc.c:
7465 * gst-libs/gst/app/gstappsrc.h:
7466 Make debug categories static. Use _element_class_set_details_simple().
7468 2009-01-06 10:56:45 +0000 Tim-Philipp Müller <tim@centricular.net>
7470 gst-libs/gst/app/: Move private data into a private instance struct. Add padding to instance and class structures exp...
7471 Original commit message from CVS:
7472 * gst-libs/gst/app/gstappsink.c: (_GstAppSinkPrivate),
7473 (gst_app_sink_class_init), (gst_app_sink_init),
7474 (gst_app_sink_dispose), (gst_app_sink_finalize),
7475 (gst_app_sink_unlock_start), (gst_app_sink_unlock_stop),
7476 (gst_app_sink_flush_unlocked), (gst_app_sink_start),
7477 (gst_app_sink_stop), (gst_app_sink_event), (gst_app_sink_preroll),
7478 (gst_app_sink_render), (gst_app_sink_getcaps),
7479 (gst_app_sink_set_caps), (gst_app_sink_get_caps),
7480 (gst_app_sink_is_eos), (gst_app_sink_set_emit_signals),
7481 (gst_app_sink_get_emit_signals), (gst_app_sink_set_max_buffers),
7482 (gst_app_sink_get_max_buffers), (gst_app_sink_set_drop),
7483 (gst_app_sink_get_drop), (gst_app_sink_pull_preroll),
7484 (gst_app_sink_pull_buffer)::
7485 * gst-libs/gst/app/gstappsink.h: (GstAppSinkPrivate), (_GstAppSink)::
7486 * gst-libs/gst/app/gstappsrc.c: (_GstAppSrcPrivate),
7487 (gst_app_src_class_init), (gst_app_src_init),
7488 (gst_app_src_flush_queued), (gst_app_src_dispose),
7489 (gst_app_src_finalize), (gst_app_src_set_property),
7490 (gst_app_src_get_property), (gst_app_src_unlock),
7491 (gst_app_src_unlock_stop), (gst_app_src_start), (gst_app_src_stop),
7492 (gst_app_src_is_seekable), (gst_app_src_check_get_range),
7493 (gst_app_src_query), (gst_app_src_do_seek), (gst_app_src_create),
7494 (gst_app_src_set_caps), (gst_app_src_get_caps),
7495 (gst_app_src_set_size), (gst_app_src_get_size),
7496 (gst_app_src_set_stream_type), (gst_app_src_get_stream_type),
7497 (gst_app_src_set_max_bytes), (gst_app_src_get_max_bytes),
7498 (gst_app_src_set_latencies), (gst_app_src_set_latency),
7499 (gst_app_src_get_latency), (gst_app_src_push_buffer_full),
7500 (gst_app_src_push_buffer_action), (gst_app_src_end_of_stream)::
7501 * gst-libs/gst/app/gstappsrc.h: (GstAppSrcPrivate)::
7502 Move private data into a private instance struct. Add padding to
7503 instance and class structures exposed in public headers. Add
7504 Since markers to the gtk-doc blurbs (#566750).
7506 2009-01-06 10:50:37 +0000 Wim Taymans <wim.taymans@gmail.com>
7508 tests/examples/app/appsrc_ex.c: Some comments.
7509 Original commit message from CVS:
7510 * tests/examples/app/appsrc_ex.c: (main):
7512 When pulling a buffer we can get NULL when the element is EOS, don't try
7513 to unref this NULL buffer.
7515 2009-01-06 10:16:16 +0000 Jan Schmidt <thaytan@mad.scientist.com>
7517 gst-libs/gst/video/: Fix up build flags and include statement for the new generated enumtypes files, to fix dist.
7518 Original commit message from CVS:
7519 * gst-libs/gst/video/Makefile.am:
7520 * gst-libs/gst/video/video.h:
7521 Fix up build flags and include statement for the new generated
7522 enumtypes files, to fix dist.
7524 2009-01-05 23:04:57 +0000 Jan Schmidt <thaytan@mad.scientist.com>
7526 Move AppSrc/AppSink from gst-plugins-bad. Fixes #564421
7527 Original commit message from CVS:
7529 * docs/libs/Makefile.am:
7530 * docs/libs/gst-plugins-base-libs-docs.sgml:
7531 * docs/libs/gst-plugins-base-libs-sections.txt:
7532 * docs/plugins/Makefile.am:
7533 * docs/plugins/gst-plugins-base-plugins-docs.sgml:
7534 * docs/plugins/gst-plugins-base-plugins-sections.txt:
7535 * docs/plugins/gst-plugins-base-plugins.args:
7536 * docs/plugins/gst-plugins-base-plugins.hierarchy:
7537 * docs/plugins/gst-plugins-base-plugins.interfaces:
7538 * docs/plugins/gst-plugins-base-plugins.prerequisites:
7539 * docs/plugins/gst-plugins-base-plugins.signals:
7540 * docs/plugins/inspect/plugin-app.xml:
7541 * gst-libs/gst/Makefile.am:
7542 * gst-libs/gst/app/gstappsink.c:
7543 * gst-libs/gst/app/gstappsrc.c:
7544 * tests/examples/Makefile.am:
7545 * tests/examples/app/Makefile.am:
7546 Move AppSrc/AppSink from gst-plugins-bad. Fixes #564421
7548 2009-01-05 17:13:13 +0000 Wim Taymans <wim.taymans@gmail.com>
7550 gst-libs/gst/audio/gstbaseaudiosink.c: Avoid holding the OBJECT_LOCK when calling ringbuffer functions that take the ...
7551 Original commit message from CVS:
7552 * gst-libs/gst/audio/gstbaseaudiosink.c:
7553 (gst_base_audio_sink_change_state):
7554 Avoid holding the OBJECT_LOCK when calling ringbuffer functions that
7555 take the ringbuffer lock because rinbuffer lock > OBJECT_LOCK. We can do
7556 this because the async_play method is deprecated and usually not called
7559 2009-01-05 12:18:52 +0000 Wim Taymans <wim.taymans@gmail.com>
7561 gst/playback/gstplaybin2.c: Disconnect signal handlers before destroying a previous decodebin so that we don't end up...
7562 Original commit message from CVS:
7563 * gst/playback/gstplaybin2.c: (notify_source_cb), (activate_group):
7564 Disconnect signal handlers before destroying a previous decodebin so
7565 that we don't end up causing deadlocks. Fixes #566586.
7567 2009-01-05 10:59:35 +0000 Wim Taymans <wim.taymans@gmail.com>
7569 gst/audiotestsrc/gstaudiotestsrc.*: Add property to control pull/push based scheduling.
7570 Original commit message from CVS:
7571 * gst/audiotestsrc/gstaudiotestsrc.c:
7572 (gst_audio_test_src_class_init), (gst_audio_test_src_init),
7573 (gst_audio_test_src_check_get_range),
7574 (gst_audio_test_src_set_property),
7575 (gst_audio_test_src_get_property):
7576 * gst/audiotestsrc/gstaudiotestsrc.h:
7577 Add property to control pull/push based scheduling.
7579 2009-01-02 15:04:13 +0000 Alessandro Decina <alessandro.d@gmail.com>
7581 Make the seek and colorkey examples depend on gtk+-x11 as they use
7582 Original commit message from CVS:
7584 * tests/examples/seek/Makefile.am:
7585 * tests/icles/Makefile.am:
7586 Make the seek and colorkey examples depend on gtk+-x11 as they use
7588 Fixes the build with gtk+-quartz.
7590 2008-12-31 16:04:26 +0000 Jan Schmidt <thaytan@mad.scientist.com>
7592 win32/common/: Add new exports to win32 files.
7593 Original commit message from CVS:
7594 * win32/common/libgstaudio.def:
7595 * win32/common/libgsttag.def:
7596 * win32/common/libgstvideo.def:
7597 Add new exports to win32 files.
7599 2008-12-31 13:31:55 +0000 Edward Hervey <bilboed@bilboed.com>
7601 gst-libs/gst/tag/gsttagdemux.*: Add GType for GstTagDemuxResult enum.
7602 Original commit message from CVS:
7603 * gst-libs/gst/tag/gsttagdemux.c: (gst_tag_demux_result_get_type):
7604 * gst-libs/gst/tag/gsttagdemux.h:
7605 Add GType for GstTagDemuxResult enum.
7607 2008-12-31 13:01:30 +0000 Edward Hervey <bilboed@bilboed.com>
7609 gst-libs/gst/video/: Add glib-mkenum for GstVideoFormat enum GTYPE auto-generation.
7610 Original commit message from CVS:
7611 * gst-libs/gst/video/Makefile.am:
7612 * gst-libs/gst/video/video.h:
7613 Add glib-mkenum for GstVideoFormat enum GTYPE auto-generation.
7614 This will help bindings to use it.
7616 2008-12-31 11:20:26 +0000 Edward Hervey <bilboed@bilboed.com>
7618 Switch glib-mkenum for gst-libs/gst/audio from multichannel- to audio- in order to wrap all enums declarations of tha...
7619 Original commit message from CVS:
7620 * gst-libs/gst/audio/Makefile.am:
7621 * gst-libs/gst/audio/audio.c:
7622 * gst-libs/gst/audio/multichannel.h:
7623 * gst-libs/gst/audio/testchannels.c:
7625 * win32/common/audio-enumtypes.c:
7626 (gst_audio_channel_position_get_type),
7627 (gst_ring_buffer_state_get_type),
7628 (gst_ring_buffer_seg_state_get_type),
7629 (gst_buffer_format_type_get_type), (gst_buffer_format_get_type):
7630 * win32/common/audio-enumtypes.h:
7631 * win32/common/multichannel-enumtypes.c:
7632 * win32/common/multichannel-enumtypes.h:
7633 * win32/vs6/grammar.dsp:
7634 * win32/vs6/libgstaudio.dsp:
7635 * win32/vs7/libgstaudio.vcproj:
7636 * win32/vs8/libgstaudio.vcproj:
7637 Switch glib-mkenum for gst-libs/gst/audio from multichannel- to
7638 audio- in order to wrap all enums declarations of that library.
7639 This modification should not matter since that header file is not a
7640 public header (it will be included by public headers).
7641 Modify win32 crap^Wfiles accordingly.
7643 2008-12-30 17:55:07 +0000 Edward Hervey <bilboed@bilboed.com>
7645 gst-libs/gst/audio/: Complete Sebastien's commit from the 13th by exporting the _slave_method_get_type() methods.
7646 Original commit message from CVS:
7647 * gst-libs/gst/audio/gstbaseaudiosrc.h:
7648 * gst-libs/gst/audio/gstbaseaudiosink.h:
7649 Complete Sebastien's commit from the 13th by exporting the
7650 _slave_method_get_type() methods.
7652 2008-12-29 16:45:20 +0000 Wim Taymans <wim.taymans@gmail.com>
7654 gst-libs/gst/app/gstappsrc.*: Add properties and methods to configure and retrieve the min and max latencies.
7655 Original commit message from CVS:
7656 * gst-libs/gst/app/gstappsrc.c: (gst_app_src_class_init),
7657 (gst_app_src_init), (gst_app_src_set_property),
7658 (gst_app_src_get_property), (gst_app_src_query),
7659 (gst_app_src_set_latencies), (gst_app_src_set_latency),
7660 (gst_app_src_get_latency), (gst_app_src_push_buffer_full):
7661 * gst-libs/gst/app/gstappsrc.h:
7662 Add properties and methods to configure and retrieve the min and max
7665 2008-12-20 17:38:41 +0000 Sebastian Dröge <slomo@circular-chaos.org>
7667 ext/: Implement URI query. Fixes bug #562949.
7668 Original commit message from CVS:
7669 * ext/gio/gstgiobasesink.c: (gst_gio_base_sink_query):
7670 * ext/gio/gstgiobasesrc.c: (gst_gio_base_src_class_init),
7671 (gst_gio_base_src_query):
7672 * ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_query):
7673 * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_class_init),
7674 (gst_gnome_vfs_src_query):
7675 Implement URI query. Fixes bug #562949.
7677 2008-12-20 12:48:43 +0000 Wim Taymans <wim.taymans@gmail.com>
7679 gst/playback/gstplaybin2.c: Add some debug info.
7680 Original commit message from CVS:
7681 * gst/playback/gstplaybin2.c: (no_more_pads_cb):
7682 Add some debug info.
7683 * gst/playback/gstplaysink.c: (gen_video_chain), (gen_audio_chain),
7684 (gst_play_sink_reconfigure), (gst_play_sink_request_pad),
7685 (gst_play_sink_release_pad):
7686 Add some more debug info.
7687 Reconfigure the audio chain when we switch between raw and encoded audio
7688 in gapless playback.
7690 2008-12-20 12:45:03 +0000 Wim Taymans <wim.taymans@gmail.com>
7692 gst-libs/gst/audio/gstbaseaudiosink.c: Pause the write thread before deactivating and releasing the ringbuffer to avo...
7693 Original commit message from CVS:
7694 * gst-libs/gst/audio/gstbaseaudiosink.c:
7695 (gst_base_audio_sink_setcaps):
7696 Pause the write thread before deactivating and releasing the ringbuffer
7697 to avoid a deadlock when we do gapless playback with different sample
7698 rates in playbin2. Fixes #564929.
7700 2008-12-19 13:03:00 +0000 Sebastian Dröge <slomo@circular-chaos.org>
7702 gst-libs/gst/audio/gstbaseaudiosrc.c: Make GstAudioSrcSlaveMethod get_type() function non-static as it's public now.
7703 Original commit message from CVS:
7704 * gst-libs/gst/audio/gstbaseaudiosrc.c:
7705 Make GstAudioSrcSlaveMethod get_type() function non-static
7707 * win32/common/libgstaudio.def:
7708 * win32/common/libgstnetbuffer.def:
7709 Add some missing functions to the list of exported symbols.
7711 2008-12-18 12:37:33 +0000 Andrew Feren <acferen@yahoo.com>
7713 gst-libs/gst/netbuffer/gstnetbuffer.*: Make gst_netaddress_get_ip4_address fail for v6 addresses.
7714 Original commit message from CVS:
7715 Patch by: Andrew Feren <acferen at yahoo dot com>
7716 * gst-libs/gst/netbuffer/gstnetbuffer.c:
7717 (gst_netaddress_get_ip4_address), (gst_netaddress_get_ip6_address),
7718 (gst_netaddress_get_address_bytes),
7719 (gst_netaddress_set_address_bytes):
7720 * gst-libs/gst/netbuffer/gstnetbuffer.h:
7721 Make gst_netaddress_get_ip4_address fail for v6 addresses.
7722 Make gst_netaddress_get_ip6_address either fail or return the v4
7723 address as a transitional v6 address.
7724 Add two convenience functions:
7725 API: gst_netaddress_get_address_bytes()
7726 API: gst_netaddress_set_address_bytes()
7729 2008-12-17 13:51:46 +0000 Wim Taymans <wim.taymans@gmail.com>
7731 Add appsrc and appsink documentation.
7732 Original commit message from CVS:
7733 * docs/plugins/Makefile.am:
7734 * docs/plugins/gst-plugins-bad-plugins-docs.sgml:
7735 * docs/plugins/gst-plugins-bad-plugins-sections.txt:
7736 * gst-libs/gst/app/gstappsink.c:
7737 * gst-libs/gst/app/gstappsrc.c: (gst_app_src_class_init):
7738 Add appsrc and appsink documentation.
7740 2008-12-17 08:51:34 +0000 Stefan Kost <ensonic@users.sourceforge.net>
7742 gst/adder/: Cleanup variable names to make the adder-loop easier to understand.
7743 Original commit message from CVS:
7744 * gst/adder/Makefile.am:
7745 * gst/adder/gstadder.c:
7746 Cleanup variable names to make the adder-loop easier to understand.
7747 Also try to use liboil to spee it up, but ifdef it out as it does not
7748 make any change for me (Intel pentim M (sse,sse2) please try on other
7751 2008-12-16 20:16:17 +0000 Wim Taymans <wim.taymans@gmail.com>
7753 Add minimal docs to make the remaining tcp elements show up.
7754 Original commit message from CVS:
7755 * docs/plugins/Makefile.am:
7756 * docs/plugins/gst-plugins-base-plugins-docs.sgml:
7757 * docs/plugins/gst-plugins-base-plugins-sections.txt:
7758 * gst/tcp/gsttcpclientsink.c:
7759 * gst/tcp/gsttcpclientsrc.c:
7760 * gst/tcp/gsttcpserversrc.c:
7761 Add minimal docs to make the remaining tcp elements show up.
7764 2008-12-15 12:02:26 +0000 Wim Taymans <wim.taymans@gmail.com>
7766 examples/app/: Fix example to unref after emiting the push-buffer action.
7767 Original commit message from CVS:
7768 * examples/app/appsrc-ra.c: (feed_data):
7769 * examples/app/appsrc-seekable.c: (feed_data):
7770 * examples/app/appsrc-stream.c: (read_data):
7771 * examples/app/appsrc-stream2.c: (feed_data):
7772 Fix example to unref after emiting the push-buffer action.
7773 * gst-libs/gst/app/gstappsrc.c: (gst_app_src_class_init),
7774 (gst_app_src_push_buffer_full), (gst_app_src_push_buffer),
7775 (gst_app_src_push_buffer_action):
7776 Don't take the ref on the buffer in push-buffer action because it's too
7777 awkward for bindings. Fixes #564482.
7779 2008-12-13 19:32:13 +0000 Tim-Philipp Müller <tim@centricular.net>
7781 win32/common/config.h: Update to CVS version.
7782 Original commit message from CVS:
7783 * win32/common/config.h:
7784 Update to CVS version.
7785 * win32/common/config.h.in:
7786 Hardcode path to plugin install helper exe, just like we hardcode
7787 the paths in core. Removes another source of VCS conflicts for
7788 people hacking gst-plugins-base on systems with autotools.
7790 2008-12-13 16:21:12 +0000 Edward Hervey <bilboed@bilboed.com>
7792 m4/Makefile.am: And a couple more .m4 that don't exist anymore with gettext 0.17
7793 Original commit message from CVS:
7795 And a couple more .m4 that don't exist anymore with gettext 0.17
7797 2008-12-13 12:41:56 +0000 Edward Hervey <bilboed@bilboed.com>
7799 m4/Makefile.am: inttypes.m4 hasn't been available since gettext-0.15, and since we now require gettext >= 0.17 ... we...
7800 Original commit message from CVS:
7802 inttypes.m4 hasn't been available since gettext-0.15, and since we now
7803 require gettext >= 0.17 ... we can remove it from the list of files to
7806 2008-12-13 06:57:09 +0000 Sebastian Dröge <slomo@circular-chaos.org>
7808 gst-libs/gst/audio/: API: Add GST_TYPE_BASE_AUDIO_(SRC|SINK)_SLAVE_METHOD to the public API. This is needed for the C...
7809 Original commit message from CVS:
7810 * gst-libs/gst/audio/gstbaseaudiosink.c:
7811 (gst_base_audio_sink_slave_method_get_type),
7812 (gst_base_audio_sink_class_init):
7813 * gst-libs/gst/audio/gstbaseaudiosink.h:
7814 * gst-libs/gst/audio/gstbaseaudiosrc.c:
7815 (gst_base_audio_src_slave_method_get_type),
7816 (gst_base_audio_src_class_init):
7817 * gst-libs/gst/audio/gstbaseaudiosrc.h:
7818 API: Add GST_TYPE_BASE_AUDIO_(SRC|SINK)_SLAVE_METHOD to the
7819 public API. This is needed for the C++ bindings to be able
7820 to use this base classes. Fixes bug #564200, #564206.
7822 2008-12-12 19:41:28 +0000 Edward Hervey <bilboed@bilboed.com>
7824 gst-libs/gst/cdda/gstcddabasesrc.c: Remove erroneous gst_buffer_ref().
7825 Original commit message from CVS:
7826 * gst-libs/gst/cdda/gstcddabasesrc.c:
7827 (gst_cdda_base_src_handle_event):
7828 Remove erroneous gst_buffer_ref().
7829 * tests/check/libs/rtp.c: (GST_START_TEST):
7830 Don't forget to unref the buffer once you're done with it.
7832 2008-12-12 13:06:48 +0000 Stefan Kost <ensonic@users.sourceforge.net>
7834 gst/playback/: XRef to GstXOverlay.
7835 Original commit message from CVS:
7836 * gst/playback/gstplaybin.c:
7837 * gst/playback/gstplaybin2.c:
7838 XRef to GstXOverlay.
7840 2008-12-12 10:54:45 +0000 Edward Hervey <bilboed@bilboed.com>
7842 gst/playback/gsturidecodebin.c: Free the factory array when finalizing.
7843 Original commit message from CVS:
7844 * gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_finalize):
7845 Free the factory array when finalizing.
7846 * sys/v4l/gstv4lsrc.c: (gst_v4lsrc_base_init):
7847 Use a GstStaticPadTemplate since the src pad caps are fixed.
7849 2008-12-12 07:17:21 +0000 Edward Hervey <bilboed@bilboed.com>
7851 ext/vorbis/vorbisenc.c: Make vorbisenc's pad template behave like vorbisdec's. Fixes a leak with pad templates.
7852 Original commit message from CVS:
7853 * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_base_init),
7854 (gst_vorbis_enc_init):
7855 Make vorbisenc's pad template behave like vorbisdec's. Fixes a leak with
7858 2008-12-12 07:15:22 +0000 Edward Hervey <bilboed@bilboed.com>
7860 gst-libs/gst/riff/riff-media.c: Add mapping for VP6 in avi/riff.
7861 Original commit message from CVS:
7862 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
7863 (gst_riff_create_video_template_caps):
7864 Add mapping for VP6 in avi/riff.
7866 2008-12-11 15:49:12 +0000 Edward Hervey <bilboed@bilboed.com>
7868 gst/subparse/samiparse.c: Some versions of libxml seem to be very picky as to strict formatting of the input and neve...
7869 Original commit message from CVS:
7870 * gst/subparse/samiparse.c: (sami_context_push_state),
7871 (sami_context_pop_state), (start_sami_element), (end_sami_element):
7872 Some versions of libxml seem to be very picky as to strict formatting
7873 of the input and never 'close' the final </body> tag.
7874 In order to fix that bad behaviour, we trigger the flushing of
7875 remaining data on both </body> and </sami>.
7878 2008-12-11 12:32:03 +0000 Guillaume Emont <guillaume@fluendo.com>
7880 gst/typefind/gsttypefindfunctions.c: Add typefinders for MS Word files and OS X .DS_Store files to prevent them to be...
7881 Original commit message from CVS:
7882 Patch by: Guillaume Emont <guillaume at fluendo dot com>
7883 * gst/typefind/gsttypefindfunctions.c: (plugin_init):
7884 Add typefinders for MS Word files and OS X .DS_Store files to
7885 prevent them to be recognized as MPEG files. Fixes bug #564098.
7887 2008-12-11 11:04:14 +0000 Wim Taymans <wim.taymans@gmail.com>
7889 gst/playback/gstplaysink.c: Add some more debug info.
7890 Original commit message from CVS:
7891 * gst/playback/gstplaysink.c: (gen_audio_chain),
7892 (gst_play_sink_reconfigure):
7893 Add some more debug info.
7894 Fix linking of just an encoded sink.
7895 Handle failure to create a sink chain more gracefully than crashing.
7897 2008-12-11 10:33:48 +0000 Wim Taymans <wim.taymans@gmail.com>
7899 tests/check/pipelines/theoraenc.c: Pushing 10 buffers is enough to run the test.
7900 Original commit message from CVS:
7901 * tests/check/pipelines/theoraenc.c: (GST_START_TEST):
7902 Pushing 10 buffers is enough to run the test.
7904 2008-12-11 10:28:43 +0000 Wim Taymans <wim.taymans@gmail.com>
7906 tests/examples/seek/seek.c: Hook up the SKIP seek flag.
7907 Original commit message from CVS:
7908 * tests/examples/seek/seek.c: (do_seek), (stop_cb),
7909 (skip_toggle_cb), (rate_spinbutton_changed_cb), (msg_segment_done),
7911 Hook up the SKIP seek flag.
7913 2008-12-10 18:43:32 +0000 Wim Taymans <wim.taymans@gmail.com>
7915 gst/playback/gstplaybin2.c: Error out with a missing-plugin error when the input-selector was not found.
7916 Original commit message from CVS:
7917 * gst/playback/gstplaybin2.c: (pad_added_cb):
7918 Error out with a missing-plugin error when the input-selector was not
7920 * gst/playback/gstplaysink.c: (gst_play_sink_reconfigure):
7923 2008-12-10 17:39:32 +0000 Wim Taymans <wim.taymans@gmail.com>
7925 gst/playback/gstplaysink.c: Use G_DEFINE_TYPE.
7926 Original commit message from CVS:
7927 * gst/playback/gstplaysink.c: (gst_play_sink_class_init),
7928 (gst_play_sink_dispose), (gst_play_sink_finalize), (try_element),
7929 (gen_video_chain), (gen_audio_chain), (gst_play_sink_reconfigure),
7930 (gst_play_sink_send_event), (gst_play_sink_change_state):
7932 Try to set the selected sink to READY before using it. This will allow
7933 for detection of incompatible formats sooner.
7934 Don't cause a fatal error when conversion elements are missing but post
7935 a missing-element message and a warning instead because things might
7936 still link and run fine.
7937 Simplyfy the construction of audio and video sink chains.
7939 2008-12-10 14:55:10 +0000 Wim Taymans <wim.taymans@gmail.com>
7941 ext/ogg/gstoggdemux.c: Use G_DEFINE_TYPE for the OggPad to get some threadsafe type init from glib.
7942 Original commit message from CVS:
7943 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_class_init),
7944 (gst_ogg_pad_dispose), (gst_ogg_pad_finalize):
7945 Use G_DEFINE_TYPE for the OggPad to get some threadsafe type
7948 2008-12-10 08:19:13 +0000 Luis Menina <liberforce@freeside.fr>
7950 gst/: Include glib.h instead of a specific GLib header. Including single
7951 Original commit message from CVS:
7952 Patch by: Luis Menina <liberforce at freeside dot fr>
7953 * gst-libs/gst/floatcast/floatcast.h:
7954 * gst/typefind/gsttypefindfunctions.c:
7955 Include glib.h instead of a specific GLib header. Including single
7956 GLib headers is deprecated. Fixes bug #563904.
7958 2008-12-09 18:30:10 +0000 Julien Moutte <julien@moutte.net>
7960 gst-libs/gst/riff/riff-media.c: Support higher max audio rates for some formats (WAV, Vorbis, LPCM).
7961 Original commit message from CVS:
7962 2008-12-09 Julien Moutte <julien@fluendo.com>
7963 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
7964 Support higher max audio rates for some formats (WAV, Vorbis, LPCM).
7966 2008-12-09 17:21:37 +0000 Stefan Kost <ensonic@users.sourceforge.net>
7968 gst-libs/gst/riff/riff-read.c: Fix handling of odd chunks in riff metadata.
7969 Original commit message from CVS:
7970 * gst-libs/gst/riff/riff-read.c:
7971 Fix handling of odd chunks in riff metadata.
7973 2008-12-08 18:44:22 +0000 Wim Taymans <wim.taymans@gmail.com>
7975 gst/volume/gstvolume.c: Use new basetransform vmethod to reconfigure the dynamic properties and any pending volume/mu...
7976 Original commit message from CVS:
7977 * gst/volume/gstvolume.c: (gst_volume_class_init),
7978 (volume_before_transform), (volume_transform_ip):
7979 Use new basetransform vmethod to reconfigure the dynamic properties and
7980 any pending volume/mute changes. Fixes #563508.
7982 2008-12-08 18:12:18 +0000 Sebastian Dröge <slomo@circular-chaos.org>
7984 configure.ac: First check for "theoraenc theoradec" and if that failed check for "theora >= 1.0alpha5". The former ap...
7985 Original commit message from CVS:
7987 First check for "theoraenc theoradec" and if that failed check
7988 for "theora >= 1.0alpha5". The former appeared in 1.0beta3 and
7989 deprecate the latter. Also linking on Windows fails with just "theora"
7990 and the version check would fail for the release candidates.
7993 2008-12-08 15:25:13 +0000 Stefan Kost <ensonic@users.sourceforge.net>
7995 gst/playback/: Add basic docs to decodebin and link to decodebin from decodebin2.
7996 Original commit message from CVS:
7997 * gst/playback/gstdecodebin.c:
7998 * gst/playback/gstdecodebin2.c:
7999 Add basic docs to decodebin and link to decodebin from decodebin2.
8001 2008-12-08 12:08:32 +0000 Olivier Crete <tester@tester.ca>
8003 gst-libs/gst/rtp/gstrtcpbuffer.*: Implement gst_rtcp_packet_remove(). Fixes #563174.
8004 Original commit message from CVS:
8005 Patch by: Olivier Crete <tester at tester ca>
8006 * gst-libs/gst/rtp/gstrtcpbuffer.c: (gst_rtcp_packet_remove):
8007 * gst-libs/gst/rtp/gstrtcpbuffer.h:
8008 Implement gst_rtcp_packet_remove(). Fixes #563174.
8009 * tests/check/libs/rtp.c: (GST_START_TEST), (rtp_suite):
8010 Add unit test for some RTCP functions.
8012 2008-12-04 20:09:19 +0000 Sebastian Dröge <slomo@circular-chaos.org>
8014 configure.ac: Apparently AC_CONFIG_MACRO_DIR breaks when using more than one macro directory, reverting last change.
8015 Original commit message from CVS:
8017 Apparently AC_CONFIG_MACRO_DIR breaks when using more
8018 than one macro directory, reverting last change.
8020 2008-12-04 19:47:12 +0000 Sebastian Dröge <slomo@circular-chaos.org>
8022 configure.ac: Set AC_CONFIG_MACRO_DIR to common/m4 to point autoconf to our M4 macros.
8023 Original commit message from CVS:
8025 Set AC_CONFIG_MACRO_DIR to common/m4 to point autoconf to
8028 2008-12-03 17:47:44 +0000 Edward Hervey <bilboed@bilboed.com>
8030 sys/: Clear all flags on buffers returned from the image pool.
8031 Original commit message from CVS:
8032 * sys/ximage/ximagesink.c: (gst_ximagesink_buffer_alloc):
8033 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_buffer_alloc):
8034 Clear all flags on buffers returned from the image pool.
8037 2008-12-01 19:36:35 +0000 이문형 <iwings@gmail.com>
8039 gst-libs/gst/app/gstappsrc.c: Don't forget to release the lock again if we bail out because some pad is flushing or w...
8040 Original commit message from CVS:
8041 Patch by: 이문형 <iwings at gmail dot com>
8042 * gst-libs/gst/app/gstappsrc.c: (gst_app_src_push_buffer):
8043 Don't forget to release the lock again if we bail out because some
8044 pad is flushing or we've reached EOS, otherwise things will lock up
8045 next time _push_buffer() is called (#562802).
8047 2008-11-29 13:31:47 +0000 Sebastian Dröge <slomo@circular-chaos.org>
8049 Require gettext 0.17 because older versions don't mix with libtool 2.2. At build time an older gettext version will s...
8050 Original commit message from CVS:
8051 Patch by: Cygwin Ports maintainer
8052 <yselkowitz at users dot sourceforge dot net>
8055 Require gettext 0.17 because older versions don't mix with libtool
8056 2.2. At build time an older gettext version will still work.
8059 2008-11-28 13:30:36 +0000 Christian Schaller <uraeus@gnome.org>
8062 * gst/speexresample/Makefile.am:
8064 Original commit message from CVS:
8067 2008-11-28 09:44:12 +0000 Sebastian Dröge <slomo@circular-chaos.org>
8069 Update documentation of speexresample for the new element name.
8070 Original commit message from CVS:
8071 * docs/plugins/gst-plugins-base-plugins.args:
8072 * docs/plugins/gst-plugins-base-plugins.hierarchy:
8073 * docs/plugins/gst-plugins-base-plugins.interfaces:
8074 * docs/plugins/gst-plugins-base-plugins.prerequisites:
8075 * docs/plugins/inspect/plugin-videorate.xml:
8076 * gst/speexresample/gstspeexresample.c:
8077 Update documentation of speexresample for the new element name.
8079 2008-11-28 09:04:46 +0000 Sebastian Dröge <slomo@circular-chaos.org>
8081 gst/speexresample/README: Update README with the latest diff between the Speex resampler and our copy.
8082 Original commit message from CVS:
8083 * gst/speexresample/README:
8084 Update README with the latest diff between the Speex resampler
8087 2008-11-28 08:37:50 +0000 Sebastian Dröge <slomo@circular-chaos.org>
8089 gst/speexresample/gstspeexresample.c: Update the debug category from speex_resample to audioresample.
8090 Original commit message from CVS:
8091 * gst/speexresample/gstspeexresample.c: (plugin_init):
8092 Update the debug category from speex_resample to audioresample.
8094 2008-11-27 19:13:59 +0000 Sebastian Dröge <slomo@circular-chaos.org>
8096 Remove audioresample files.
8097 Original commit message from CVS:
8098 * gst/audioresample/Makefile.am:
8099 * gst/audioresample/buffer.c:
8100 * gst/audioresample/buffer.h:
8101 * gst/audioresample/debug.c:
8102 * gst/audioresample/debug.h:
8103 * gst/audioresample/functable.c:
8104 * gst/audioresample/functable.h:
8105 * gst/audioresample/gstaudioresample.c:
8106 * gst/audioresample/gstaudioresample.h:
8107 * gst/audioresample/resample.c:
8108 * gst/audioresample/resample.h:
8109 * gst/audioresample/resample_chunk.c:
8110 * gst/audioresample/resample_functable.c:
8111 * gst/audioresample/resample_ref.c:
8112 * tests/check/elements/audioresample.c:
8113 Remove audioresample files.
8115 2008-11-27 17:04:07 +0000 Sebastian Dröge <slomo@circular-chaos.org>
8117 docs/plugins/inspect/plugin-audioresample.xml: Regenerated for library filename change.
8118 Original commit message from CVS:
8119 * docs/plugins/inspect/plugin-audioresample.xml:
8120 Regenerated for library filename change.
8122 2008-11-27 16:57:09 +0000 Sebastian Dröge <slomo@circular-chaos.org>
8124 Rename the moved speexresample to audioresample, integrate into the build system and remove the old audioresample fro...
8125 Original commit message from CVS:
8127 * docs/plugins/Makefile.am:
8128 * docs/plugins/gst-plugins-base-plugins-sections.txt:
8129 * docs/plugins/gst-plugins-base-plugins.args:
8130 * docs/plugins/gst-plugins-base-plugins.hierarchy:
8131 * docs/plugins/gst-plugins-base-plugins.interfaces:
8132 * docs/plugins/gst-plugins-base-plugins.prerequisites:
8133 * docs/plugins/inspect/plugin-adder.xml:
8134 * docs/plugins/inspect/plugin-alsa.xml:
8135 * docs/plugins/inspect/plugin-audioconvert.xml:
8136 * docs/plugins/inspect/plugin-audiorate.xml:
8137 * docs/plugins/inspect/plugin-audioresample.xml:
8138 * docs/plugins/inspect/plugin-audiotestsrc.xml:
8139 * docs/plugins/inspect/plugin-cdparanoia.xml:
8140 * docs/plugins/inspect/plugin-decodebin.xml:
8141 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
8142 * docs/plugins/inspect/plugin-gdp.xml:
8143 * docs/plugins/inspect/plugin-gio.xml:
8144 * docs/plugins/inspect/plugin-gnomevfs.xml:
8145 * docs/plugins/inspect/plugin-libvisual.xml:
8146 * docs/plugins/inspect/plugin-ogg.xml:
8147 * docs/plugins/inspect/plugin-pango.xml:
8148 * docs/plugins/inspect/plugin-playback.xml:
8149 * docs/plugins/inspect/plugin-queue2.xml:
8150 * docs/plugins/inspect/plugin-subparse.xml:
8151 * docs/plugins/inspect/plugin-tcp.xml:
8152 * docs/plugins/inspect/plugin-theora.xml:
8153 * docs/plugins/inspect/plugin-typefindfunctions.xml:
8154 * docs/plugins/inspect/plugin-uridecodebin.xml:
8155 * docs/plugins/inspect/plugin-video4linux.xml:
8156 * docs/plugins/inspect/plugin-videorate.xml:
8157 * docs/plugins/inspect/plugin-videoscale.xml:
8158 * docs/plugins/inspect/plugin-videotestsrc.xml:
8159 * docs/plugins/inspect/plugin-volume.xml:
8160 * docs/plugins/inspect/plugin-vorbis.xml:
8161 * docs/plugins/inspect/plugin-ximagesink.xml:
8162 * docs/plugins/inspect/plugin-xvimagesink.xml:
8163 * gst/speexresample/gstspeexresample.c: (plugin_init):
8164 * gst/speexresample/Makefile.am:
8165 * tests/check/Makefile.am:
8166 * tests/check/elements/speexresample.c: (setup_speexresample),
8167 (GST_START_TEST), (test_pipeline):
8168 Rename the moved speexresample to audioresample, integrate into the
8169 build system and remove the old audioresample from the build system.
8170 Fixes bug #558124, #385061, #346218, #116051.
8172 2008-11-27 16:47:41 +0000 Wim Taymans <wim.taymans@gmail.com>
8174 gst-libs/gst/audio/gstbaseaudiosrc.c: Avoid nasty int overflows after about 12 hours and 25 minutes when these code p...
8175 Original commit message from CVS:
8176 * gst-libs/gst/audio/gstbaseaudiosrc.c:
8177 (gst_base_audio_src_get_offset), (gst_base_audio_src_create):
8178 Avoid nasty int overflows after about 12 hours and 25 minutes when these
8179 code paths are triggered.
8180 A free beer to Håvard Graff for finding this!
8182 2008-11-27 11:16:44 +0000 이문형 <iwings@gmail.com>
8184 gst-libs/gst/rtsp/gstrtspconnection.c: A successful gst_poll_wait() doesn't always mean successful connect() on
8185 Original commit message from CVS:
8186 Patch by: 이문형 <iwings at gmail dot com>
8187 * gst-libs/gst/rtsp/gstrtspconnection.c:
8188 (gst_rtsp_connection_connect):
8189 A successful gst_poll_wait() doesn't always mean successful connect() on
8190 Windows. We should check errors by calling gst_poll_fd_has_error().
8193 2008-11-25 16:37:50 +0000 Sebastian Dröge <slomo@circular-chaos.org>
8195 tests/check/elements/speexresample.c: Make unit test again faster to prevent timeouts with valgrind.
8196 Original commit message from CVS:
8197 * tests/check/elements/speexresample.c: (test_pipeline):
8198 Make unit test again faster to prevent timeouts with valgrind.
8200 2008-11-25 15:33:30 +0000 Wim Taymans <wim.taymans@gmail.com>
8202 gst-libs/gst/rtp/gstrtcpbuffer.c: Fix typo in the docs.
8203 Original commit message from CVS:
8204 * gst-libs/gst/rtp/gstrtcpbuffer.c:
8205 Fix typo in the docs.
8207 2008-11-25 15:28:36 +0000 Wim Taymans <wim.taymans@gmail.com>
8209 ext/ogg/gstoggdemux.c: If no stream was found before receiving EOS, post an error message.
8210 Original commit message from CVS:
8211 * ext/ogg/gstoggdemux.c: (gst_ogg_demux_sink_event):
8212 If no stream was found before receiving EOS, post an error message.
8215 2008-11-25 15:14:30 +0000 Wim Taymans <wim.taymans@gmail.com>
8217 ext/theora/: Parse segment events.
8218 Original commit message from CVS:
8219 * ext/theora/gsttheoraenc.h:
8220 * ext/theora/theoraenc.c: (gst_theora_enc_init),
8221 (theora_buffer_from_packet), (theora_push_packet),
8222 (theora_enc_sink_event), (theora_enc_is_discontinuous),
8224 Parse segment events.
8225 Pass incomming buffer timestamps to outgoing buffers.
8226 Use the running_time to construct the granulepos.
8229 2008-11-25 11:00:55 +0000 Wim Taymans <wim.taymans@gmail.com>
8231 gst/playback/gstplaybin2.c: Fix buffer-duration property.
8232 Original commit message from CVS:
8233 * gst/playback/gstplaybin2.c: (activate_group):
8234 Fix buffer-duration property.
8236 2008-11-25 10:32:49 +0000 Wim Taymans <wim.taymans@gmail.com>
8238 gst-libs/gst/audio/gstbaseaudiosink.c: Really fix audiosink drain handling by keeping track of the running_time of th...
8239 Original commit message from CVS:
8240 * gst-libs/gst/audio/gstbaseaudiosink.c:
8241 (gst_base_audio_sink_drain), (gst_base_audio_sink_event),
8242 (gst_base_audio_sink_sync_latency), (gst_base_audio_sink_render),
8243 (gst_base_audio_sink_change_state):
8244 Really fix audiosink drain handling by keeping track of the running_time
8247 2008-11-24 20:25:24 +0000 Michael Smith <msmith@xiph.org>
8249 gst/playback/gstplaybin2.c: Add notification of current stream. Add ability to configure buffer sizes.
8250 Original commit message from CVS:
8251 * gst/playback/gstplaybin2.c:
8252 Add notification of current stream. Add ability to configure buffer
8254 * gst/playback/gsturidecodebin.c:
8255 Add ability to configure buffer sizes for streaming mode.
8258 2008-11-24 20:11:52 +0000 Stefan Kost <ensonic@users.sourceforge.net>
8260 gst-libs/gst/audio/gstbaseaudiosink.c: Time is already in running_time. Remove base_time handling. Fixes audiosinks n...
8261 Original commit message from CVS:
8262 * gst-libs/gst/audio/gstbaseaudiosink.c:
8263 Time is already in running_time. Remove base_time handling. Fixes
8264 audiosinks not draining and thus chopping some audio in the end.
8266 2008-11-24 19:18:59 +0000 David Schleef <ds@schleef.org>
8268 ext/ogg/gstoggmux.*: If we're muxing a dirac stream, flush the page after every picture.
8269 Original commit message from CVS:
8270 * ext/ogg/gstoggmux.c:
8271 * ext/ogg/gstoggmux.h:
8272 If we're muxing a dirac stream, flush the page after every picture.
8274 2008-11-24 12:56:54 +0000 Stefan Kost <ensonic@users.sourceforge.net>
8276 gst-libs/gst/audio/gstbaseaudiosink.c: Add one log message to check for audio_drained. Sync one log message with the ...
8277 Original commit message from CVS:
8278 * gst-libs/gst/audio/gstbaseaudiosink.c:
8279 Add one log message to check for audio_drained. Sync one log message
8280 with the condition. Send EOS after draining audio in pull mode.
8282 2008-11-24 12:07:10 +0000 Sebastian Dröge <slomo@circular-chaos.org>
8284 ext/: Use gst_buffer_try_new_and_alloc() and fail properly if the allocation failed. This prevents abort() if downstr...
8285 Original commit message from CVS:
8286 * ext/gio/gstgiobasesrc.c: (gst_gio_base_src_create):
8287 * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_create):
8288 Use gst_buffer_try_new_and_alloc() and fail properly if the
8289 allocation failed. This prevents abort() if downstream elements
8290 request an insane amount of memory.
8292 2008-11-24 12:03:11 +0000 Jon Trowbridge <trow@ximian.com>
8294 gst/volume/gstvolume.*: Cleanup volume, define and use default values.
8295 Original commit message from CVS:
8296 * gst/volume/gstvolume.c: (volume_choose_func),
8297 (volume_update_volume), (gst_volume_set_volume),
8298 (gst_volume_get_volume), (gst_volume_set_mute),
8299 (gst_volume_class_init), (gst_volume_init),
8300 (volume_process_double), (volume_process_float),
8301 (volume_process_int32), (volume_process_int32_clamp),
8302 (volume_process_int24), (volume_process_int24_clamp),
8303 (volume_process_int16), (volume_process_int16_clamp),
8304 (volume_process_int8), (volume_process_int8_clamp), (volume_setup),
8305 (volume_transform_ip), (volume_set_property),
8306 (volume_get_property):
8307 * gst/volume/gstvolume.h:
8308 Cleanup volume, define and use default values.
8309 Recalculate new volume and mute setup before processing. Fixes #561789.
8310 * tests/check/elements/volume.c: (GST_START_TEST), (volume_suite):
8311 Add controller unit test. Patch by: Jonathan Matthew
8312 Fix bogus test that messed with basetransform's internal state.
8314 2008-11-22 15:02:15 +0000 Sebastian Dröge <slomo@circular-chaos.org>
8316 tests/check/elements/speexresample.c: Make the unit test a bit faster to prevent timeouts, especially with valgrind.
8317 Original commit message from CVS:
8318 * tests/check/elements/speexresample.c: (GST_START_TEST):
8319 Make the unit test a bit faster to prevent timeouts, especially
8322 2008-11-22 14:44:26 +0000 Wim Taymans <wim.taymans@gmail.com>
8324 gst/videorate/gstvideorate.c: Add jpeg and png image media types to the caps. Fixes #561436.
8325 Original commit message from CVS:
8326 * gst/videorate/gstvideorate.c:
8327 Add jpeg and png image media types to the caps. Fixes #561436.
8329 2008-11-22 14:31:43 +0000 Wim Taymans <wim.taymans@gmail.com>
8331 gst/playback/gstplaysink.c: Don't post an error when we can't configure the volume but post a warning instead. Fixes ...
8332 Original commit message from CVS:
8333 * gst/playback/gstplaysink.c: (gen_audio_chain):
8334 Don't post an error when we can't configure the volume but post a
8335 warning instead. Fixes #561780.
8337 2008-11-21 20:32:56 +0000 Jonathan Rosser <jonathan.rosser@rd.bbc.co.uk>
8339 gst/videotestsrc/: Add a zone plate pattern generator based on BBC R&D Report 1978/23 (yeah *that* 1978). Try 'video...
8340 Original commit message from CVS:
8341 Patch by: Jonathan Rosser <jonathan.rosser@rd.bbc.co.uk>
8342 * gst/videotestsrc/gstvideotestsrc.c:
8343 * gst/videotestsrc/gstvideotestsrc.h:
8344 * gst/videotestsrc/videotestsrc.c:
8345 * gst/videotestsrc/videotestsrc.h:
8346 Add a zone plate pattern generator based on BBC R&D Report
8347 1978/23 (yeah *that* 1978). Try 'videotestsrc pattern=zone-plate
8348 kx2=20 ky2=20 kt=1'.
8350 2008-11-21 15:45:15 +0000 Sebastian Dröge <slomo@circular-chaos.org>
8352 gst/speexresample/gstspeexresample.c: Add a "filter-length" property that maps to the quality values for compatibilty...
8353 Original commit message from CVS:
8354 * gst/speexresample/gstspeexresample.c:
8355 (gst_speex_resample_class_init), (gst_speex_resample_set_property),
8356 (gst_speex_resample_get_property):
8357 Add a "filter-length" property that maps to the quality values
8358 for compatibilty with audioresample.
8360 2008-11-21 00:04:48 +0000 Michael Smith <msmith@xiph.org>
8362 gst/playback/gstdecodebin2.c: Fix random fat-fingering making this not compile.
8363 Original commit message from CVS:
8364 * gst/playback/gstdecodebin2.c:
8365 Fix random fat-fingering making this not compile.
8367 2008-11-20 22:11:38 +0000 Michael Smith <msmith@xiph.org>
8369 gst/playback/gstdecodebin2.c: If the top-level type of the stream is plain text, don't try to decode it, matching beh...
8370 Original commit message from CVS:
8371 * gst/playback/gstdecodebin2.c:
8372 If the top-level type of the stream is plain text, don't try to decode
8373 it, matching behaviour of decodebin.
8374 * gst/playback/gstplaysink.c:
8375 If we fail to generate a text chain (e.g. due to missing optional
8376 plugins), don't crash.
8378 2008-11-20 22:06:05 +0000 Michael Smith <msmith@xiph.org>
8380 gst-libs/gst/rtsp/gstrtspdefs.c: Fix win32 build. Oops.
8381 Original commit message from CVS:
8382 * gst-libs/gst/rtsp/gstrtspdefs.c:
8383 Fix win32 build. Oops.
8385 2008-11-20 21:40:49 +0000 Michael Smith <msmith@xiph.org>
8387 gst-libs/gst/rtsp/gstrtspdefs.c: Use WSAGetLastError() rather than errno/h_errno on win32.
8388 Original commit message from CVS:
8389 * gst-libs/gst/rtsp/gstrtspdefs.c:
8390 Use WSAGetLastError() rather than errno/h_errno on win32.
8392 2008-11-20 21:20:27 +0000 Michael Smith <msmith@xiph.org>
8394 gst-libs/gst/riff/riff-media.c: Support WMA Lossless properly.
8395 Original commit message from CVS:
8396 * gst-libs/gst/riff/riff-media.c:
8397 Support WMA Lossless properly.
8399 2008-11-19 00:24:44 +0000 David Schleef <ds@schleef.org>
8401 gst/videotestsrc/: Add "colorspec" property, specifying whether to generate BT.601 or BT.709 video. This only affect...
8402 Original commit message from CVS:
8403 * gst/videotestsrc/gstvideotestsrc.c:
8404 * gst/videotestsrc/gstvideotestsrc.h:
8405 * gst/videotestsrc/videotestsrc.c:
8406 * gst/videotestsrc/videotestsrc.h:
8407 Add "colorspec" property, specifying whether to generate BT.601
8408 or BT.709 video. This only affects YCbCr values, not RGB, since
8409 if you're generating a 709 test pattern, presumably you want
8410 709 RGB primaries, not 601. Also add "smpte75" pattern, which
8411 uses 75% colors instead of 100%, since this is often more useful
8412 for testing (and also follows the SMPTE EG-1 guideline).
8414 2008-11-18 18:08:42 +0000 Alessandro Decina <alessandro.d@gmail.com>
8416 gst/playback/gstdecodebin.c: Add a "sink-caps" property to decodebin like it's done for decodebin2.
8417 Original commit message from CVS:
8418 * gst/playback/gstdecodebin.c:
8419 Add a "sink-caps" property to decodebin like it's done for decodebin2.
8422 2008-11-14 21:44:33 +0000 Jan Schmidt <thaytan@mad.scientist.com>
8424 gst/audioresample/gstaudioresample.c: Guard against a NULL dereference I somehow encountered - with a FLUSH_STOP arri...
8425 Original commit message from CVS:
8426 * gst/audioresample/gstaudioresample.c:
8427 Guard against a NULL dereference I somehow encountered -
8428 with a FLUSH_STOP arriving either before basetransform _start(),
8430 * gst/typefind/gsttypefindfunctions.c:
8431 Make sure we never jump backwards when typefinding corrupt mov files.
8433 2008-11-14 21:39:09 +0000 Jan Schmidt <thaytan@mad.scientist.com>
8435 gst-libs/gst/interfaces/propertyprobe.c: Fix random type causing a docs warning.
8436 Original commit message from CVS:
8437 * gst-libs/gst/interfaces/propertyprobe.c:
8438 Fix random type causing a docs warning.
8440 2008-11-14 15:40:28 +0000 Stefan Kost <ensonic@users.sourceforge.net>
8442 sys/v4l/gstv4l.c: Give it a minimal rank for autovideosrc.
8443 Original commit message from CVS:
8445 Give it a minimal rank for autovideosrc.
8447 2008-11-13 21:11:13 +0000 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
8449 gst/typefind/gsttypefindfunctions.c: Improve typefinding of ISO JPEG2000 mime types.
8450 Original commit message from CVS:
8451 * gst/typefind/gsttypefindfunctions.c: (jp2_type_find),
8453 Improve typefinding of ISO JPEG2000 mime types.
8455 2008-11-13 18:18:32 +0000 Wim Taymans <wim.taymans@gmail.com>
8457 sys/xvimage/xvimagesink.*: Avoid typechecking when we do trivial casts.
8458 Original commit message from CVS:
8459 * sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_finalize),
8460 (gst_xvimagesink_xvimage_put), (gst_xvimagesink_setcaps),
8461 (gst_xvimagesink_show_frame), (gst_xvimagesink_buffer_alloc):
8462 * sys/xvimage/xvimagesink.h:
8463 Avoid typechecking when we do trivial casts.
8464 Move error handling out of the main program flow.
8465 Sneak in the display-region caps property, not completely correct yet.
8466 Cache the width/height in buffer_alloc instead of parsing it from the
8469 2008-11-13 17:27:37 +0000 Wim Taymans <wim.taymans@gmail.com>
8471 gst/playback/gstplaybin2.c: don't try to unlink the selector sinkpad when we don't have it yet. This can happen if an...
8472 Original commit message from CVS:
8473 * gst/playback/gstplaybin2.c: (deactivate_group):
8474 don't try to unlink the selector sinkpad when we don't have it yet. This
8475 can happen if an error occured before the group was complete.
8477 2008-11-13 15:37:40 +0000 Wim Taymans <wim.taymans@gmail.com>
8479 gst-libs/gst/rtp/gstrtpbuffer.c: Avoid expensive type checks we already did as part of the _validate() function that ...
8480 Original commit message from CVS:
8481 * gst-libs/gst/rtp/gstrtpbuffer.c: (gst_rtp_buffer_validate_data),
8482 (gst_rtp_buffer_set_packet_len), (gst_rtp_buffer_get_packet_len),
8483 (gst_rtp_buffer_get_header_len), (gst_rtp_buffer_get_version),
8484 (gst_rtp_buffer_set_version), (gst_rtp_buffer_get_padding),
8485 (gst_rtp_buffer_set_padding), (gst_rtp_buffer_pad_to),
8486 (gst_rtp_buffer_get_extension), (gst_rtp_buffer_set_extension),
8487 (gst_rtp_buffer_get_extension_data),
8488 (gst_rtp_buffer_set_extension_data), (gst_rtp_buffer_get_ssrc),
8489 (gst_rtp_buffer_set_ssrc), (gst_rtp_buffer_get_csrc_count),
8490 (gst_rtp_buffer_get_csrc), (gst_rtp_buffer_set_csrc),
8491 (gst_rtp_buffer_get_marker), (gst_rtp_buffer_set_marker),
8492 (gst_rtp_buffer_get_payload_type),
8493 (gst_rtp_buffer_set_payload_type), (gst_rtp_buffer_get_seq),
8494 (gst_rtp_buffer_set_seq), (gst_rtp_buffer_get_timestamp),
8495 (gst_rtp_buffer_set_timestamp),
8496 (gst_rtp_buffer_get_payload_subbuffer),
8497 (gst_rtp_buffer_get_payload_len), (gst_rtp_buffer_get_payload):
8498 Avoid expensive type checks we already did as part of the
8499 _validate() function that should be called first.
8501 2008-11-11 16:40:50 +0000 Wim Taymans <wim.taymans@gmail.com>
8503 gst-libs/gst/rtp/gstbasertpdepayload.c: Fix some cases where a newsegment event was not sent.
8504 Original commit message from CVS:
8505 * gst-libs/gst/rtp/gstbasertpdepayload.c: (create_segment_event),
8506 (gst_base_rtp_depayload_push_full),
8507 (gst_base_rtp_depayload_set_gst_timestamp):
8508 Fix some cases where a newsegment event was not sent.
8510 2008-11-11 15:52:14 +0000 Wim Taymans <wim.taymans@gmail.com>
8512 gst/playback/gstplaybin2.c: Catch state change errors and stop from the uridecodebin elements instead of trying to co...
8513 Original commit message from CVS:
8514 * gst/playback/gstplaybin2.c: (activate_group):
8515 Catch state change errors and stop from the uridecodebin elements
8516 instead of trying to continue in vain.
8518 2008-11-10 14:53:45 +0000 Edward Hervey <bilboed@bilboed.com>
8520 gst/: Wim, you're a bad boy. You don't want people to contact you or what?
8521 Original commit message from CVS:
8522 * gst-libs/gst/app/gstappsink.c:
8523 * gst-libs/gst/app/gstappsrc.c:
8524 * gst/h264parse/gsth264parse.c:
8525 Wim, you're a bad boy. You don't want people to contact you or what?
8527 2008-11-10 14:22:09 +0000 Wim Taymans <wim.taymans@gmail.com>
8529 gst-libs/gst/audio/gstbaseaudiosink.c: Use gst_base_sink_do_preroll() to wait for PLAYING and before waiting for the ...
8530 Original commit message from CVS:
8531 * gst-libs/gst/audio/gstbaseaudiosink.c:
8532 (gst_base_audio_sink_sync_latency), (gst_base_audio_sink_render),
8533 (gst_base_audio_sink_callback):
8534 Use gst_base_sink_do_preroll() to wait for PLAYING and before waiting
8535 for the latency to expire, fixes #559567.
8537 2008-11-10 13:55:08 +0000 Thomas Vander Stichele <thomas@apestaart.org>
8539 gst/adder/gstadder.c: Change author string after seeing output of gst-inspector.
8540 Original commit message from CVS:
8541 * gst/adder/gstadder.c:
8542 Change author string after seeing output of gst-inspector.
8544 2008-11-10 10:33:26 +0000 Wim Taymans <wim.taymans@gmail.com>
8546 gst/playback/gstplaysink.c: Don't try to do crazy things when we only have a text pad without a video pad. Fixes #559...
8547 Original commit message from CVS:
8548 * gst/playback/gstplaysink.c: (gst_play_sink_reconfigure):
8549 Don't try to do crazy things when we only have a text pad without a
8550 video pad. Fixes #559478.
8552 2008-11-07 17:35:46 +0000 Wim Taymans <wim.taymans@gmail.com>
8554 gst-libs/gst/app/gstappsrc.*: Add is-live property.
8555 Original commit message from CVS:
8556 * gst-libs/gst/app/gstappsrc.c: (gst_app_src_class_init),
8557 (gst_app_src_init), (gst_app_src_set_property),
8558 (gst_app_src_get_property), (gst_app_src_push_buffer):
8559 * gst-libs/gst/app/gstappsrc.h:
8560 Add is-live property.
8563 2008-11-06 12:14:51 +0000 Wim Taymans <wim.taymans@gmail.com>
8565 gst-libs/gst/riff/riff-media.c: Fix case where we don't have a range for the rates or channels as is the case with tr...
8566 Original commit message from CVS:
8567 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
8568 Fix case where we don't have a range for the rates or channels as is the
8569 case with truespeech.
8571 2008-11-05 19:18:25 +0000 Wim Taymans <wim.taymans@gmail.com>
8573 gst/volume/gstvolume.*: Keep negotiated state in a separate variable.
8574 Original commit message from CVS:
8575 * gst/volume/gstvolume.c: (volume_update_real_volume),
8576 (gst_volume_set_volume), (gst_volume_get_volume),
8577 (gst_volume_set_mute), (gst_volume_init), (volume_setup),
8578 (volume_transform_ip), (volume_update_mute),
8579 (volume_update_volume), (volume_get_property):
8580 * gst/volume/gstvolume.h:
8581 Keep negotiated state in a separate variable.
8582 Protect the volume and mute properties with the object lock.
8583 Protect modifying the transform with the transform lock.
8585 2008-11-05 12:20:21 +0000 Wim Taymans <wim.taymans@gmail.com>
8587 gst/ffmpegcolorspace/gstffmpegcodecmap.c: Only convert caps to string when debug is enabled.
8588 Original commit message from CVS:
8589 * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
8590 (gst_ffmpeg_pixfmt_to_caps):
8591 Only convert caps to string when debug is enabled.
8593 2008-11-04 18:17:24 +0000 Wim Taymans <wim.taymans@gmail.com>
8595 ext/theora/: Copy seqnum.
8596 Original commit message from CVS:
8597 * ext/theora/gsttheoradec.h:
8598 * ext/theora/theoradec.c: (gst_theora_dec_init),
8599 (gst_theora_dec_reset), (theora_dec_src_event),
8600 (theora_dec_sink_event), (theora_handle_type_packet):
8602 Keep events in a pending list, like vorbisdec, instead of trying
8603 to construct a segment event ourselves.
8604 * ext/vorbis/vorbisdec.c: (gst_vorbis_dec_reset),
8605 (vorbis_dec_src_event), (vorbis_dec_sink_event):
8606 * ext/vorbis/vorbisdec.h:
8609 2008-11-04 17:24:35 +0000 Wim Taymans <wim.taymans@gmail.com>
8611 ext/ogg/gstoggdemux.*: Copy seqnums around to track playback segments and messages.
8612 Original commit message from CVS:
8613 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_packet),
8614 (gst_ogg_demux_deactivate_current_chain),
8615 (gst_ogg_demux_perform_seek), (gst_ogg_demux_handle_page),
8616 (gst_ogg_demux_loop):
8617 * ext/ogg/gstoggdemux.h:
8618 Copy seqnums around to track playback segments and messages.
8620 2008-11-04 12:42:18 +0000 Stefan Kost <ensonic@users.sourceforge.net>
8622 Don't install static libs for plugins. Fixes #550851 for -bad.
8623 Original commit message from CVS:
8624 * ext/alsaspdif/Makefile.am:
8625 * ext/amrwb/Makefile.am:
8626 * ext/apexsink/Makefile.am:
8627 * ext/arts/Makefile.am:
8628 * ext/artsd/Makefile.am:
8629 * ext/audiofile/Makefile.am:
8630 * ext/audioresample/Makefile.am:
8631 * ext/bz2/Makefile.am:
8632 * ext/cdaudio/Makefile.am:
8633 * ext/celt/Makefile.am:
8634 * ext/dc1394/Makefile.am:
8635 * ext/dirac/Makefile.am:
8636 * ext/directfb/Makefile.am:
8637 * ext/divx/Makefile.am:
8638 * ext/dts/Makefile.am:
8639 * ext/faac/Makefile.am:
8640 * ext/faad/Makefile.am:
8641 * ext/gsm/Makefile.am:
8642 * ext/hermes/Makefile.am:
8643 * ext/ivorbis/Makefile.am:
8644 * ext/jack/Makefile.am:
8645 * ext/jp2k/Makefile.am:
8646 * ext/ladspa/Makefile.am:
8647 * ext/lcs/Makefile.am:
8648 * ext/libfame/Makefile.am:
8649 * ext/libmms/Makefile.am:
8650 * ext/metadata/Makefile.am:
8651 * ext/mpeg2enc/Makefile.am:
8652 * ext/mplex/Makefile.am:
8653 * ext/musepack/Makefile.am:
8654 * ext/musicbrainz/Makefile.am:
8655 * ext/mythtv/Makefile.am:
8656 * ext/nas/Makefile.am:
8657 * ext/neon/Makefile.am:
8658 * ext/ofa/Makefile.am:
8659 * ext/polyp/Makefile.am:
8660 * ext/resindvd/Makefile.am:
8661 * ext/sdl/Makefile.am:
8662 * ext/shout/Makefile.am:
8663 * ext/snapshot/Makefile.am:
8664 * ext/sndfile/Makefile.am:
8665 * ext/soundtouch/Makefile.am:
8666 * ext/spc/Makefile.am:
8667 * ext/swfdec/Makefile.am:
8668 * ext/tarkin/Makefile.am:
8669 * ext/theora/Makefile.am:
8670 * ext/timidity/Makefile.am:
8671 * ext/twolame/Makefile.am:
8672 * ext/x264/Makefile.am:
8673 * ext/xine/Makefile.am:
8674 * ext/xvid/Makefile.am:
8675 * gst-libs/gst/app/Makefile.am:
8676 * gst-libs/gst/dshow/Makefile.am:
8677 * gst/aiffparse/Makefile.am:
8678 * gst/app/Makefile.am:
8679 * gst/audiobuffer/Makefile.am:
8680 * gst/bayer/Makefile.am:
8681 * gst/cdxaparse/Makefile.am:
8682 * gst/chart/Makefile.am:
8683 * gst/colorspace/Makefile.am:
8684 * gst/dccp/Makefile.am:
8685 * gst/deinterlace/Makefile.am:
8686 * gst/deinterlace2/Makefile.am:
8687 * gst/dvdspu/Makefile.am:
8688 * gst/festival/Makefile.am:
8689 * gst/filter/Makefile.am:
8690 * gst/flacparse/Makefile.am:
8691 * gst/flv/Makefile.am:
8692 * gst/games/Makefile.am:
8693 * gst/h264parse/Makefile.am:
8694 * gst/librfb/Makefile.am:
8695 * gst/mixmatrix/Makefile.am:
8696 * gst/modplug/Makefile.am:
8697 * gst/mpeg1sys/Makefile.am:
8698 * gst/mpeg4videoparse/Makefile.am:
8699 * gst/mpegdemux/Makefile.am:
8700 * gst/mpegtsmux/Makefile.am:
8701 * gst/mpegvideoparse/Makefile.am:
8702 * gst/mve/Makefile.am:
8703 * gst/nsf/Makefile.am:
8704 * gst/nuvdemux/Makefile.am:
8705 * gst/overlay/Makefile.am:
8706 * gst/passthrough/Makefile.am:
8707 * gst/pcapparse/Makefile.am:
8708 * gst/playondemand/Makefile.am:
8709 * gst/rawparse/Makefile.am:
8710 * gst/real/Makefile.am:
8711 * gst/rtjpeg/Makefile.am:
8712 * gst/rtpmanager/Makefile.am:
8713 * gst/scaletempo/Makefile.am:
8714 * gst/sdp/Makefile.am:
8715 * gst/selector/Makefile.am:
8716 * gst/smooth/Makefile.am:
8717 * gst/smoothwave/Makefile.am:
8718 * gst/speed/Makefile.am:
8719 * gst/speexresample/Makefile.am:
8720 * gst/stereo/Makefile.am:
8721 * gst/subenc/Makefile.am:
8722 * gst/tta/Makefile.am:
8723 * gst/vbidec/Makefile.am:
8724 * gst/videodrop/Makefile.am:
8725 * gst/videosignal/Makefile.am:
8726 * gst/virtualdub/Makefile.am:
8727 * gst/vmnc/Makefile.am:
8728 * gst/y4m/Makefile.am:
8729 * sys/acmenc/Makefile.am:
8730 * sys/cdrom/Makefile.am:
8731 * sys/dshowdecwrapper/Makefile.am:
8732 * sys/dshowsrcwrapper/Makefile.am:
8733 * sys/dvb/Makefile.am:
8734 * sys/dxr3/Makefile.am:
8735 * sys/fbdev/Makefile.am:
8736 * sys/oss4/Makefile.am:
8737 * sys/qcam/Makefile.am:
8738 * sys/qtwrapper/Makefile.am:
8739 * sys/vcd/Makefile.am:
8740 * sys/wininet/Makefile.am:
8741 * win32/common/config.h:
8742 Don't install static libs for plugins. Fixes #550851 for -bad.
8744 2008-11-03 15:30:14 +0000 Matthias Kretz <kretz@kde.org>
8746 ext/alsa/gstalsasink.c: Make all access non-blocking so that we can better handle unplugging of usb devices. Fixes #5...
8747 Original commit message from CVS:
8748 Based on patch by: Matthias Kretz <kretz at kde dot org>
8749 * ext/alsa/gstalsasink.c: (gst_alsasink_open),
8750 (gst_alsasink_prepare), (gst_alsasink_unprepare),
8751 (gst_alsasink_write):
8752 Make all access non-blocking so that we can better handle unplugging
8753 of usb devices. Fixes #559111
8755 2008-11-03 10:49:24 +0000 Damien Lespiau <damien.lespiau@gmail.com>
8757 gst-libs/gst/rtsp/gstrtspconnection.c: Make the next call to poll not depend on previous calls to poll with or withou...
8758 Original commit message from CVS:
8759 Patch by: Damien Lespiau <damien.lespiau gmail com>
8760 * gst-libs/gst/rtsp/gstrtspconnection.c:
8761 (gst_rtsp_connection_write):
8762 Make the next call to poll not depend on previous calls to poll with or
8763 without reading from the active descriptor. Fixes #544293.
8765 2008-11-03 08:55:49 +0000 Sebastian Dröge <slomo@circular-chaos.org>
8767 gst/speexresample/gstspeexresample.c: Add TODO at the top of the file for enabling SSE/ARM specific optimizations and...
8768 Original commit message from CVS:
8769 * gst/speexresample/gstspeexresample.c:
8770 (gst_speex_resample_convert_buffer):
8771 Add TODO at the top of the file for enabling SSE/ARM specific
8772 optimizations and choosing the fastest implementation at runtime.
8773 Add g_assert_not_reached() at two places that should really never
8776 2008-11-02 09:19:24 +0000 Sebastian Dröge <slomo@circular-chaos.org>
8778 gst/speexresample/gstspeexresample.c: Fix format string and arguments.
8779 Original commit message from CVS:
8780 * gst/speexresample/gstspeexresample.c:
8781 (gst_speex_resample_check_discont):
8782 Fix format string and arguments.
8783 * gst/speexresample/resample_sse.h:
8786 2008-11-01 19:38:36 +0000 Sebastian Dröge <slomo@circular-chaos.org>
8788 gst/speexresample/: Add missing headers to Makefile.am.
8789 Original commit message from CVS:
8790 * gst/speexresample/Makefile.am:
8791 * gst/speexresample/gstspeexresample.c:
8792 (gst_speex_resample_base_init), (gst_speex_resample_get_funcs),
8793 (gst_speex_resample_convert_buffer), (_benchmark_int_float),
8794 (_benchmark_int_int), (_benchmark_integer_resampling),
8796 * gst/speexresample/gstspeexresample.h:
8797 * gst/speexresample/resample.c:
8798 * gst/speexresample/speex_resampler_double.c:
8799 * gst/speexresample/speex_resampler_float.c:
8800 * gst/speexresample/speex_resampler_int.c:
8801 * gst/speexresample/speex_resampler_wrapper.h:
8802 Add missing headers to Makefile.am.
8803 Update copyright, years and my mail address.
8804 Benchmark the integer resampling implementation against the
8805 float implementation and use the faster one for 8/16 bit integer
8806 input. On most recent systems the floating point version is faster.
8808 2008-10-31 09:49:57 +0000 Nick Haddad <nick@haddads.net>
8810 gst-libs/gst/riff/: Add support for other fourcc codes that are commonly used for 'uncompressed RGB', including 'RGB ...
8811 Original commit message from CVS:
8812 Patch by: Nick Haddad <nick at haddads dot net>
8813 * gst-libs/gst/riff/riff-ids.h:
8814 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
8815 Add support for other fourcc codes that are commonly used for
8816 'uncompressed RGB', including 'RGB ', 'RAW ', and 0.
8819 2008-10-30 14:55:43 +0000 Sebastian Dröge <slomo@circular-chaos.org>
8821 gst/speexresample/gstspeexresample.c: The length for the buffer conversion function is the number of audio frames, i....
8822 Original commit message from CVS:
8823 * gst/speexresample/gstspeexresample.c:
8824 (gst_speex_resample_convert_buffer):
8825 The length for the buffer conversion function is the number of
8826 audio frames, i.e. we need to multiply it by the number of channels
8827 to get the number of values. Also spotted by the unit test after
8828 running in valgrind.
8830 2008-10-30 14:46:31 +0000 Sebastian Dröge <slomo@circular-chaos.org>
8832 tests/check/elements/speexresample.c: Add pipeline unit tests for testing all supported formats with up/downsampling ...
8833 Original commit message from CVS:
8834 * tests/check/elements/speexresample.c: (element_message_cb),
8835 (eos_message_cb), (test_pipeline), (GST_START_TEST),
8836 (speexresample_suite):
8837 Add pipeline unit tests for testing all supported formats with
8838 up/downsampling and different in/outrates.
8839 * gst/speexresample/gstspeexresample.c:
8840 (gst_speex_resample_push_drain), (gst_speex_resample_process):
8841 * gst/speexresample/speex_resampler_wrapper.h:
8842 Fix bugs identified by the testsuite.
8844 2008-10-30 13:44:41 +0000 Sebastian Dröge <slomo@circular-chaos.org>
8846 gst/speexresample/: Add support for int8, int24 and int32 input by converting internally to/from int16 or double.
8847 Original commit message from CVS:
8848 * gst/speexresample/gstspeexresample.c: (gst_speex_resample_stop),
8849 (gst_speex_resample_get_funcs),
8850 (gst_speex_resample_transform_size),
8851 (gst_speex_resample_convert_buffer),
8852 (gst_speex_resample_push_drain), (gst_speex_resample_process):
8853 * gst/speexresample/gstspeexresample.h:
8854 * gst/speexresample/speex_resampler_wrapper.h:
8855 Add support for int8, int24 and int32 input by converting internally
8856 to/from int16 or double.
8858 2008-10-30 12:43:44 +0000 Sebastian Dröge <slomo@circular-chaos.org>
8860 Add support for double samples as input and refactor the usage of the different compilation flavors of the speex resa...
8861 Original commit message from CVS:
8862 * gst/speexresample/Makefile.am:
8863 * gst/speexresample/arch.h:
8864 * gst/speexresample/gstspeexresample.c: (gst_speex_resample_stop),
8865 (gst_speex_resample_get_unit_size), (gst_speex_resample_get_funcs),
8866 (gst_speex_resample_init_state), (gst_speex_resample_update_state),
8867 (gst_speex_resample_reset_state), (gst_speex_resample_parse_caps),
8868 (_gcd), (gst_speex_resample_transform_size),
8869 (gst_speex_resample_set_caps), (gst_speex_resample_push_drain),
8870 (gst_speex_resample_process), (gst_speex_resample_transform),
8871 (gst_speex_resample_query), (gst_speex_resample_set_property):
8872 * gst/speexresample/gstspeexresample.h:
8873 * gst/speexresample/resample.c:
8874 * gst/speexresample/speex_resampler.h:
8875 * gst/speexresample/speex_resampler_double.c:
8876 * gst/speexresample/speex_resampler_wrapper.h:
8877 * tests/check/elements/speexresample.c: (setup_speexresample),
8878 (test_perfect_stream_instance), (GST_START_TEST),
8879 (test_discont_stream_instance):
8880 Add support for double samples as input and refactor the usage
8881 of the different compilation flavors of the speex resampler.
8883 2008-10-30 11:43:12 +0000 Stefan Kost <ensonic@users.sourceforge.net>
8885 gst/audioresample/gstaudioresample.c: Return the result of parent_class->event().
8886 Original commit message from CVS:
8887 * gst/audioresample/gstaudioresample.c:
8888 Return the result of parent_class->event().
8890 2008-10-29 17:02:55 +0000 Wim Taymans <wim.taymans@gmail.com>
8892 gst-libs/gst/app/gstappsink.c: Fix the docs.
8893 Original commit message from CVS:
8894 * gst-libs/gst/app/gstappsink.c: (gst_app_sink_class_init):
8897 2008-10-29 12:11:20 +0000 Sebastian Dröge <slomo@circular-chaos.org>
8899 gst/speexresample/gstspeexresample.*: Rewrite timestamp tracking to make it more robust and guarantee a continous str...
8900 Original commit message from CVS:
8901 * gst/speexresample/gstspeexresample.c: (gst_speex_resample_start),
8902 (gst_speex_resample_get_unit_size),
8903 (gst_speex_resample_push_drain), (gst_speex_resample_event),
8904 (gst_speex_resample_check_discont), (gst_speex_resample_process),
8905 (gst_speex_resample_transform):
8906 * gst/speexresample/gstspeexresample.h:
8907 Rewrite timestamp tracking to make it more robust and guarantee
8909 * tests/check/Makefile.am:
8910 * tests/check/elements/speexresample.c: (setup_speexresample),
8911 (cleanup_speexresample), (fail_unless_perfect_stream),
8912 (test_perfect_stream_instance), (GST_START_TEST),
8913 (test_discont_stream_instance), (live_switch_alloc_only_48000),
8914 (live_switch_get_sink_caps), (live_switch_push),
8915 (speexresample_suite):
8916 Add unit tests for speexresample based on the audioresample unit tests.
8918 2008-10-28 19:30:33 +0000 Sebastian Dröge <slomo@circular-chaos.org>
8920 gst/speexresample/gstspeexresample.*: Some random cleanup, add G_LIKELY and friends, use GST_DEBUG_OBJECT instead of ...
8921 Original commit message from CVS:
8922 * gst/speexresample/gstspeexresample.c:
8923 (gst_speex_resample_get_unit_size),
8924 (gst_speex_resample_fixate_caps), (gst_speex_resample_init_state),
8925 (gst_speex_resample_update_state), (gst_speex_resample_parse_caps),
8926 (gst_speex_resample_transform_size), (gst_speex_resample_set_caps),
8927 (gst_speex_resample_push_drain), (gst_speex_resample_event),
8928 (gst_speex_resample_check_discont), (gst_speex_fix_output_buffer),
8929 (gst_speex_resample_process), (gst_speex_resample_transform),
8930 (gst_speex_resample_query), (gst_speex_resample_set_property):
8931 * gst/speexresample/gstspeexresample.h:
8932 Some random cleanup, add G_LIKELY and friends, use GST_DEBUG_OBJECT
8933 instead of GST_DEBUG, ...
8935 2008-10-28 16:28:45 +0000 Sebastian Dröge <slomo@circular-chaos.org>
8937 gst/speexresample/gstspeexresample.c: Fixate to the nearest supported rate instead of the first one.
8938 Original commit message from CVS:
8939 * gst/speexresample/gstspeexresample.c:
8940 (gst_speex_resample_class_init), (gst_speex_resample_fixate_caps),
8941 (gst_speex_resample_process):
8942 Fixate to the nearest supported rate instead of the first one.
8944 2008-10-28 16:25:00 +0000 Sebastian Dröge <slomo@circular-chaos.org>
8946 gst/audioresample/gstaudioresample.c: Fixate the rate to the nearest supported rate instead of the first one. Fixes b...
8947 Original commit message from CVS:
8948 * gst/audioresample/gstaudioresample.c:
8949 (gst_audioresample_class_init), (audioresample_fixate_caps):
8950 Fixate the rate to the nearest supported rate instead of
8951 the first one. Fixes bug #549510.
8953 2008-10-28 11:46:28 +0000 Sebastian Dröge <slomo@circular-chaos.org>
8955 gst/speexresample/: Update Speex resampler with latest version from Speex GIT.
8956 Original commit message from CVS:
8957 * gst/speexresample/README:
8958 * gst/speexresample/arch.h:
8959 * gst/speexresample/fixed_arm4.h:
8960 * gst/speexresample/fixed_arm5e.h:
8961 * gst/speexresample/fixed_bfin.h:
8962 * gst/speexresample/fixed_debug.h:
8963 * gst/speexresample/fixed_generic.h:
8964 * gst/speexresample/resample.c: (compute_func), (main), (sinc),
8965 (cubic_coef), (resampler_basic_direct_single),
8966 (resampler_basic_direct_double),
8967 (resampler_basic_interpolate_single),
8968 (resampler_basic_interpolate_double), (update_filter),
8969 (speex_resampler_init_frac), (speex_resampler_process_native),
8970 (speex_resampler_magic), (speex_resampler_process_float),
8971 (speex_resampler_process_int),
8972 (speex_resampler_process_interleaved_float),
8973 (speex_resampler_process_interleaved_int),
8974 (speex_resampler_set_rate_frac), (speex_resampler_skip_zeros),
8975 (speex_resampler_reset_mem):
8976 * gst/speexresample/speex_resampler.h:
8977 Update Speex resampler with latest version from Speex GIT.
8979 2008-10-27 14:57:34 +0000 Wim Taymans <wim.taymans@gmail.com>
8981 win32/common/libgstaudio.def: Add new symbols.
8982 Original commit message from CVS:
8983 * win32/common/libgstaudio.def:
8986 2008-10-23 09:57:06 +0000 Wim Taymans <wim.taymans@gmail.com>
8988 ext/vorbis/vorbisdec.c: Attempt to make obfuscated code clearer.
8989 Original commit message from CVS:
8990 * ext/vorbis/vorbisdec.c: (vorbis_handle_identification_packet):
8991 Attempt to make obfuscated code clearer.
8993 2008-10-23 07:11:23 +0000 Sebastian Dröge <slomo@circular-chaos.org>
8995 Move float endianness conversion macros to core. Second part of bug ##555196.
8996 Original commit message from CVS:
8997 * docs/libs/gst-plugins-base-libs-sections.txt:
8998 * gst-libs/gst/floatcast/floatcast.h:
8999 Move float endianness conversion macros to core. Second part of
9002 2008-10-22 12:29:30 +0000 Stefan Kost <ensonic@users.sourceforge.net>
9004 sys/: Don't mark as gtk-doc docs as they aren't public.
9005 Original commit message from CVS:
9006 * sys/ximage/ximagesink.h:
9007 * sys/xvimage/xvimagesink.h:
9008 Don't mark as gtk-doc docs as they aren't public.
9010 2008-10-22 12:25:02 +0000 Stefan Kost <ensonic@users.sourceforge.net>
9012 Allow setting colorkey if possible. Implement property probe interface for optional X features (autopaint-colorkey, d...
9013 Original commit message from CVS:
9014 * sys/xvimage/xvimagesink.c:
9015 * sys/xvimage/xvimagesink.h:
9016 * tests/icles/Makefile.am:
9017 * tests/icles/test-colorkey.c:
9018 Allow setting colorkey if possible. Implement property probe interface
9019 for optional X features (autopaint-colorkey, double-buffer and
9020 colorkey). Fixes #554533
9022 2008-10-22 12:01:32 +0000 Sebastian Dröge <slomo@circular-chaos.org>
9024 gst-libs/gst/tag/tags.c: Remove useless buffer size assignment. It already has this value.
9025 Original commit message from CVS:
9026 * gst-libs/gst/tag/tags.c: (gst_tag_image_data_to_image_buffer):
9027 Remove useless buffer size assignment. It already has this value.
9029 2008-10-20 15:35:37 +0000 Wim Taymans <wim.taymans@gmail.com>
9031 gst-libs/gst/audio/gstaudiosink.c: Implement a separate activate functions to start monitoring the segments or, in pu...
9032 Original commit message from CVS:
9033 * gst-libs/gst/audio/gstaudiosink.c:
9034 (gst_audioringbuffer_class_init), (gst_audioringbuffer_acquire),
9035 (gst_audioringbuffer_activate), (gst_audioringbuffer_release),
9036 (gst_audioringbuffer_stop):
9037 Implement a separate activate functions to start monitoring the segments
9038 or, in pull mode, pulling in data.
9039 * gst-libs/gst/audio/gstbaseaudiosink.c:
9040 (gst_base_audio_sink_init), (gst_base_audio_sink_dispose),
9041 (gst_base_audio_sink_query_pad), (gst_base_audio_sink_query),
9042 (gst_base_audio_sink_setcaps), (gst_base_audio_sink_callback),
9043 (gst_base_audio_sink_activate_pull),
9044 (gst_base_audio_sink_async_play),
9045 (gst_base_audio_sink_change_state):
9046 Implement pad and element convert query function.
9047 Activate the ringbuffer.
9048 Use the segment last_stop value as the offset to pull.
9049 Use new basesink _do_preroll() method to preroll in the pulling thread.
9050 Take appropriate locking in the pulling thread.
9051 * gst-libs/gst/audio/gstringbuffer.h:
9054 2008-10-20 14:08:52 +0000 Sebastian Dröge <slomo@circular-chaos.org>
9056 gst/typefind/gsttypefindfunctions.c: Improve MXF typefinding a bit by searching for a header partition pack instead o...
9057 Original commit message from CVS:
9058 * gst/typefind/gsttypefindfunctions.c: (mxf_type_find):
9059 Improve MXF typefinding a bit by searching for a header partition
9060 pack instead of just a general partition pack and checking more
9061 bytes for valid values.
9063 2008-10-20 13:45:55 +0000 Wim Taymans <wim.taymans@gmail.com>
9065 tests/icles/.cvsignore: update ignore file.
9066 Original commit message from CVS:
9067 * tests/icles/.cvsignore:
9069 * tests/icles/Makefile.am:
9070 * tests/icles/test-box.c: (make_pipeline), (main):
9071 Add another interactive command line experimentation suite for
9072 dynamically boxing/cropping/saling an input video.
9074 2008-10-17 13:19:05 +0000 Wim Taymans <wim.taymans@gmail.com>
9076 Add methods to more accuratly control the pulling thread of a ringbuffer.
9077 Original commit message from CVS:
9078 * docs/libs/gst-plugins-base-libs-sections.txt:
9079 * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_convert),
9080 (gst_ring_buffer_activate), (gst_ring_buffer_is_active):
9081 * gst-libs/gst/audio/gstringbuffer.h:
9082 Add methods to more accuratly control the pulling thread of a
9084 Add format conversion helper code to the ringbuffer.
9085 API: GstRingBuffer:gst_ring_buffer_activate()
9086 API: GstRingBuffer:gst_ring_buffer_is_active()
9087 API: GstRingBuffer:gst_ring_buffer_convert()
9089 2008-10-16 15:44:37 +0000 Wim Taymans <wim.taymans@gmail.com>
9091 gst-libs/gst/audio/gstaudiosink.c: Signal thread startup earlier so that we can immediatly go into pull mode when we ...
9092 Original commit message from CVS:
9093 * gst-libs/gst/audio/gstaudiosink.c: (audioringbuffer_thread_func),
9094 (gst_audioringbuffer_acquire), (gst_audioringbuffer_release),
9095 (gst_audioringbuffer_stop):
9096 Signal thread startup earlier so that we can immediatly go into pull
9097 mode when we have to and block on preroll.
9099 2008-10-16 15:38:50 +0000 Wim Taymans <wim.taymans@gmail.com>
9101 gst-libs/gst/audio/gstringbuffer.c: In pull mode we want the callback to prepull a buffer we can preroll on even when...
9102 Original commit message from CVS:
9103 * gst-libs/gst/audio/gstringbuffer.c:
9104 (gst_ring_buffer_prepare_read):
9105 In pull mode we want the callback to prepull a buffer we can preroll on
9106 even when we are not yet playing.
9108 2008-10-16 15:07:00 +0000 Stefan Kost <ensonic@users.sourceforge.net>
9110 Don't install static libs for plugins. Fixes #550851 for base.
9111 Original commit message from CVS:
9112 * ext/alsa/Makefile.am:
9113 * ext/cdparanoia/Makefile.am:
9114 * ext/gio/Makefile.am:
9115 * ext/gnomevfs/Makefile.am:
9116 * ext/libvisual/Makefile.am:
9117 * ext/ogg/Makefile.am:
9118 * ext/pango/Makefile.am:
9119 * ext/theora/Makefile.am:
9120 * ext/vorbis/Makefile.am:
9121 * gst/adder/Makefile.am:
9122 * gst/audioconvert/Makefile.am:
9123 * gst/audiorate/Makefile.am:
9124 * gst/audioresample/Makefile.am:
9125 * gst/audiotestsrc/Makefile.am:
9126 * gst/ffmpegcolorspace/Makefile.am:
9127 * gst/gdp/Makefile.am:
9128 * gst/playback/Makefile.am:
9129 * gst/subparse/Makefile.am:
9130 * gst/tcp/Makefile.am:
9131 * gst/typefind/Makefile.am:
9132 * gst/videorate/Makefile.am:
9133 * gst/videoscale/Makefile.am:
9134 * gst/videotestsrc/Makefile.am:
9135 * gst/volume/Makefile.am:
9136 * sys/v4l/Makefile.am:
9137 * sys/ximage/Makefile.am:
9138 * sys/xvimage/Makefile.am:
9139 Don't install static libs for plugins. Fixes #550851 for base.
9141 2008-10-16 13:50:00 +0000 Wim Taymans <wim.taymans@gmail.com>
9143 gst/audiotestsrc/gstaudiotestsrc.c: Set the default blocksize to -1 because we will then use the configured samplespe...
9144 Original commit message from CVS:
9145 * gst/audiotestsrc/gstaudiotestsrc.c: (gst_audio_test_src_init):
9146 Set the default blocksize to -1 because we will then use the configured
9147 samplesperbuffer to create our output buffer.
9149 2008-10-15 15:28:41 +0000 Edward Hervey <bilboed@bilboed.com>
9151 gst-libs/gst/riff/riff-media.c: Add mappping for the KMVC (Karl Morton's Video) Codec.
9152 Original commit message from CVS:
9153 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
9154 (gst_riff_create_video_template_caps):
9155 Add mappping for the KMVC (Karl Morton's Video) Codec.
9157 2008-10-15 14:25:50 +0000 Edward Hervey <bilboed@bilboed.com>
9159 gst/typefind/gsttypefindfunctions.c: Don't forget to advance the offset of what we're matching against, else we end u...
9160 Original commit message from CVS:
9161 * gst/typefind/gsttypefindfunctions.c: (flac_type_find):
9162 Don't forget to advance the offset of what we're matching against, else
9163 we end up in a forever loop.
9165 2008-10-15 11:25:09 +0000 Sebastian Dröge <slomo@circular-chaos.org>
9167 gst/subparse/gstsubparse.c: Improve typefinding a bit. If we don't have a Unicode charset try GST_SUBTITLE_ENCODING a...
9168 Original commit message from CVS:
9169 * gst/subparse/gstsubparse.c: (gst_subparse_type_find):
9170 Improve typefinding a bit. If we don't have a Unicode charset
9171 try GST_SUBTITLE_ENCODING and otherwise try ISO-8859-15.
9173 2008-10-14 11:13:59 +0000 Edward Hervey <bilboed@bilboed.com>
9175 ext/theora/theoradec.c: Fix build on macosx.
9176 Original commit message from CVS:
9177 * ext/theora/theoradec.c: (theora_dec_decode_buffer):
9178 Fix build on macosx.
9180 2008-10-13 11:36:13 +0000 Robin Stocker <robin@nibor.org>
9182 ext/theora/: Parse input caps and make the PAR override the encoded PAR when specified by a container. Fixes #555699.
9183 Original commit message from CVS:
9184 Based on patch by: Robin Stocker <robin at nibor dot org>
9185 * ext/theora/gsttheoradec.h:
9186 * ext/theora/theoradec.c: (gst_theora_dec_init),
9187 (theora_dec_setcaps), (theora_handle_type_packet),
9188 (theora_dec_decode_buffer), (theora_dec_change_state):
9189 Parse input caps and make the PAR override the encoded PAR when
9190 specified by a container. Fixes #555699.
9192 2008-10-13 09:16:59 +0000 Wim Taymans <wim.taymans@gmail.com>
9194 gst-libs/gst/rtp/gstbasertpdepayload.*: Add some more G_LIKELY
9195 Original commit message from CVS:
9196 * gst-libs/gst/rtp/gstbasertpdepayload.c:
9197 (gst_base_rtp_depayload_setcaps), (gst_base_rtp_depayload_chain),
9198 (gst_base_rtp_depayload_set_gst_timestamp),
9199 (gst_base_rtp_depayload_change_state):
9200 * gst-libs/gst/rtp/gstbasertpdepayload.h:
9201 Add some more G_LIKELY
9202 Fail when the setcaps function was not called.
9203 * gst-libs/gst/rtp/gstbasertppayload.c:
9204 (gst_basertppayload_set_outcaps):
9205 Propagate return value of setcaps.
9207 2008-10-13 08:58:29 +0000 Sebastian Dröge <slomo@circular-chaos.org>
9209 gst/subparse/: Add support for UTF16/UTF32 subtitles as long as the first bytes of the first buffer contain the BOM. ...
9210 Original commit message from CVS:
9211 * gst/subparse/Makefile.am:
9212 * gst/subparse/gstsubparse.c: (gst_sub_parse_dispose),
9213 (gst_sub_parse_class_init), (gst_sub_parse_init),
9214 (gst_convert_to_utf8), (detect_encoding), (convert_encoding),
9215 (get_next_line), (gst_sub_parse_data_format_autodetect),
9216 (feed_textbuf), (handle_buffer), (gst_sub_parse_change_state),
9217 (gst_subparse_type_find):
9218 * gst/subparse/gstsubparse.h:
9219 Add support for UTF16/UTF32 subtitles as long as the first bytes of
9220 the first buffer contain the BOM. This also adds support for other
9221 encodings that allow NUL bytes via the encoding property.
9222 Fixes bugs #552237 and #456788.
9224 2008-10-13 08:15:13 +0000 Sebastian Dröge <slomo@circular-chaos.org>
9226 gst-libs/gst/tag/tags.c: Don't drop the last byte of image tags if they're not an URI list.
9227 Original commit message from CVS:
9228 * gst-libs/gst/tag/tags.c: (gst_tag_image_data_to_image_buffer):
9229 Don't drop the last byte of image tags if they're not an URI list.
9232 2008-10-13 08:00:55 +0000 Sebastian Dröge <slomo@circular-chaos.org>
9234 gst/typefind/gsttypefindfunctions.c: For looking at the 4th byte we have to get 4 bytes of course and not 3.
9235 Original commit message from CVS:
9236 * gst/typefind/gsttypefindfunctions.c: (flac_type_find):
9237 For looking at the 4th byte we have to get 4 bytes of course
9240 2008-10-13 07:52:41 +0000 Sebastian Dröge <slomo@circular-chaos.org>
9242 gst/typefind/gsttypefindfunctions.c: Improve FLAC-without-headers typefinding by looking at most of the frame header ...
9243 Original commit message from CVS:
9244 * gst/typefind/gsttypefindfunctions.c: (flac_type_find):
9245 Improve FLAC-without-headers typefinding by looking at most of the
9246 frame header and checking if invalid values are used. Should prevent
9247 quite some false positives compared to the old version which only
9248 check if the first 14 bits are set.
9250 2008-10-11 16:27:28 +0000 Stefan Kost <ensonic@users.sourceforge.net>
9252 sys/xvimage/xvimagesink.c: Don't assert on caps==NULL.
9253 Original commit message from CVS:
9254 * sys/xvimage/xvimagesink.c:
9255 Don't assert on caps==NULL.
9257 2008-10-10 17:13:40 +0000 Sebastian Dröge <slomo@circular-chaos.org>
9259 Add support for subtitle files with UTF-8 BOM at the beginning by simple stripping it from the first line before pass...
9260 Original commit message from CVS:
9261 * gst/subparse/gstsubparse.c:
9262 (gst_sub_parse_data_format_autodetect), (handle_buffer),
9263 (gst_sub_parse_change_state):
9264 * gst/subparse/gstsubparse.h:
9265 * tests/check/elements/subparse.c: (GST_START_TEST):
9266 Add support for subtitle files with UTF-8 BOM at the beginning
9267 by simple stripping it from the first line before passing it
9268 to any parsing code. Fixes bug #555257 and playback of files
9269 created by Gnome Subtitles.
9271 2008-10-10 15:45:15 +0000 Wim Taymans <wim.taymans@gmail.com>
9273 gst/audiotestsrc/gstaudiotestsrc.*: Define the default property values in the usual place.
9274 Original commit message from CVS:
9275 * gst/audiotestsrc/gstaudiotestsrc.c:
9276 (gst_audio_test_src_class_init), (gst_audio_test_src_init),
9277 (gst_audio_test_src_src_fixate), (gst_audio_test_src_setcaps),
9278 (gst_audio_test_src_start), (gst_audio_test_src_stop),
9279 (gst_audio_test_src_do_seek), (gst_audio_test_src_check_get_range),
9280 (gst_audio_test_src_create):
9281 * gst/audiotestsrc/gstaudiotestsrc.h:
9282 Define the default property values in the usual place.
9283 Implement start/stop to reset values correctly.
9284 Calculate the sample size only once when we negotiate.
9285 Rename some values to make more sense.
9286 Keep track of our byte range.
9287 Add support for pull based scheduling. Disabled for now until we have
9288 the whole stack working.
9289 Set the BUFFER_OFFSET correctly.
9291 2008-10-10 15:32:10 +0000 Sebastian Dröge <slomo@circular-chaos.org>
9293 Make the detection of the used subtitle a bit less strict for srt subtitles. Fixes bug #555607.
9294 Original commit message from CVS:
9295 Based on a patch by: xavierb at gmail dot com
9296 * gst/subparse/gstsubparse.c:
9297 (gst_sub_parse_data_format_autodetect):
9298 * tests/check/elements/subparse.c: (GST_START_TEST):
9299 Make the detection of the used subtitle a bit less strict
9300 for srt subtitles. Fixes bug #555607.
9302 2008-10-10 15:21:38 +0000 Sebastian Dröge <slomo@circular-chaos.org>
9304 ext/vorbis/vorbisenc.c: Fix discontinuity detection which was broken by last commit.
9305 Original commit message from CVS:
9306 * ext/vorbis/vorbisenc.c:
9307 (gst_vorbis_enc_buffer_check_discontinuous):
9308 Fix discontinuity detection which was broken by last commit.
9310 2008-10-09 11:18:09 +0000 Tim-Philipp Müller <tim@centricular.net>
9312 configure.ac: Require core CVS for ghostpad API additions used by decodebin2.
9313 Original commit message from CVS:
9315 Require core CVS for ghostpad API additions used by decodebin2.
9317 2008-10-08 15:30:33 +0000 Edward Hervey <bilboed@bilboed.com>
9319 gst-libs/gst/audio/gstbaseaudiosrc.c: Fix debug statements (space between '%' and actual format).
9320 Original commit message from CVS:
9321 * gst-libs/gst/audio/gstbaseaudiosrc.c:
9322 (gst_base_audio_src_create):
9323 Fix debug statements (space between '%' and actual format).
9325 2008-10-08 14:44:04 +0000 Wim Taymans <wim.taymans@gmail.com>
9327 gst/playback/gstdecodebin2.c: Remove bogus assert, the decodepad could have been created inside an already existing g...
9328 Original commit message from CVS:
9329 * gst/playback/gstdecodebin2.c: (gst_decode_pad_activate):
9330 Remove bogus assert, the decodepad could have been created inside an
9331 already existing group.
9333 2008-10-08 14:01:42 +0000 Andy Wingo <wingo@pobox.com>
9337 Original commit message from CVS:
9340 2008-10-08 14:00:07 +0000 Andy Wingo <wingo@pobox.com>
9342 gst/playback/gstdecodebin2.c (expose_pad): Fix typo: unset target instead of setting it.
9343 Original commit message from CVS:
9344 2008-10-08 Andy Wingo <wingo@pobox.com>
9345 * gst/playback/gstdecodebin2.c (expose_pad): Fix typo: unset
9346 target instead of setting it.
9347 (gst_decode_pad_activate, gst_decode_pad_unblock): This is now the
9348 API for a decode pad. The bugfix is that we set the group in
9349 activate(), not when the pad was created because it might be NULL
9351 (gst_decode_group_control_source_pad, gst_decode_group_expose):
9352 Update to use the API.
9354 2008-10-08 12:49:40 +0000 Andy Wingo <wingo@pobox.com>
9356 gst/playback/gstdecodebin2.c (struct _GstDecodePad): Change to be a subclass of GstGhostPad.
9357 Original commit message from CVS:
9358 2008-10-08 Andy Wingo <wingo@pobox.com>
9359 * gst/playback/gstdecodebin2.c (struct _GstDecodePad): Change to
9360 be a subclass of GstGhostPad.
9361 (analyze_new_pad): So, when emitting the signals that determine
9362 how we do autoplugging, already create the ghost pad and use it as
9363 the pad in the signal arguments. This allows applications to make
9364 a connection between the pad passed in e.g. autoplug-continue, and
9365 the pad passed in new-decoded-pad.
9366 (connect_pad, expose_pad): Update to receive the ghosted decode
9367 pad in the args, retargetting it as necessary if we have to plug
9368 the target pad through a multiqueue.
9369 (gst_decode_group_control_source_pad): Adapt to receive an
9370 already-ghosted pad that just needs activation, blocking, and
9372 (sort_end_pads): Adapt for decode pads actually being pads.
9373 (gst_decode_group_expose): Adapt for decode pads actually being
9374 pads. Rewrite the decode pad names so they appear in order. Adds a
9375 new error case if we couldn't set the name.
9376 (gst_decode_group_free, gst_decode_group_hide): Adapt cleanup
9378 (gst_decode_pad_set_blocked, gst_decode_pad_add_drained_check):
9379 New API for the decode pad, needed because we shouldn't do these
9380 things inside gst_decode_pad_new(), but after.
9381 (gst_decode_pad_new): Change to actually make the real pad, and
9382 delay the blocking/drainage bits.
9384 2008-10-08 12:12:01 +0000 Daniel Drake <dsd@laptop.org>
9386 ext/ogg/gstoggmux.c: Unref all buffers when clearing collectpads. Fixes bug #546955.
9387 Original commit message from CVS:
9388 Patch by: Daniel Drake <dsd at laptop dot org>
9389 * ext/ogg/gstoggmux.c: (gst_ogg_mux_clear_collectpads):
9390 Unref all buffers when clearing collectpads. Fixes bug #546955.
9392 2008-10-08 12:08:01 +0000 Klaas <klaas@rivercrew.net>
9394 ext/vorbis/vorbisenc.*: Keep track of the upstream segments and use the running time on that segment instead of the b...
9395 Original commit message from CVS:
9396 Based on a patch by: Klaas <klaas at rivercrew dot net>
9397 * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_sink_event),
9398 (gst_vorbis_enc_buffer_check_discontinuous),
9399 (gst_vorbis_enc_chain), (gst_vorbis_enc_change_state):
9400 * ext/vorbis/vorbisenc.h:
9401 Keep track of the upstream segments and use the running time on that
9402 segment instead of the buffer timestamp everywhere. Fixes bug #525807.
9404 2008-10-08 11:50:50 +0000 Sebastian Dröge <slomo@circular-chaos.org>
9406 gst/audioconvert/audioconvert.c: Prevent overflows with big buffer when calculating the size of the intermediate buff...
9407 Original commit message from CVS:
9408 * gst/audioconvert/audioconvert.c: (audio_convert_convert):
9409 Prevent overflows with big buffer when calculating the size of
9410 the intermediate buffer by using gst_util_uint64_scale() instead of
9411 plain arithmetics. Fixes bug #552801.
9413 2008-10-08 10:49:15 +0000 Pavel Zeldin <pzeldin@gmail.com>
9415 ext/pango/gstclockoverlay.*: API: Add ability to specify format for date/time display by adding a "time-format" prope...
9416 Original commit message from CVS:
9417 Patch by: Pavel Zeldin <pzeldin at gmail dot com>
9418 * ext/pango/gstclockoverlay.c: (gst_clock_overlay_render_time),
9419 (gst_clock_overlay_class_init), (gst_clock_overlay_finalize),
9420 (gst_clock_overlay_init), (gst_clock_overlay_set_property),
9421 (gst_clock_overlay_get_property):
9422 * ext/pango/gstclockoverlay.h:
9423 API: Add ability to specify format for date/time display by
9424 adding a "time-format" property.
9427 2008-10-08 09:22:26 +0000 Jan Gerber <j@oil21.org>
9429 gst-libs/gst/riff/riff-media.c: Add FFV1 fourcc to support playback of FFMPEG lossless video in AVI. Fixes bug #555319.
9430 Original commit message from CVS:
9431 Patch by: Jan Gerber <j at oil21 dot org>
9432 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
9433 (gst_riff_create_video_template_caps):
9434 Add FFV1 fourcc to support playback of FFMPEG lossless video
9435 in AVI. Fixes bug #555319.
9437 2008-10-08 09:12:36 +0000 Håvard Graff <havard.graff@tandberg.com>
9439 gst-libs/gst/audio/gstbaseaudiosrc.c: Implement skew clock slaving. Fixes #552559.
9440 Original commit message from CVS:
9441 Patch by: Håvard Graff <havard dot graff at tandberg dot com>
9442 * gst-libs/gst/audio/gstbaseaudiosrc.c:
9443 (gst_base_audio_src_create):
9444 Implement skew clock slaving. Fixes #552559.
9446 2008-10-08 09:10:23 +0000 Wim Taymans <wim.taymans@gmail.com>
9448 gst-libs/gst/audio/: Fix include of config.h
9449 Original commit message from CVS:
9450 * gst-libs/gst/audio/multichannel.c:
9451 * gst-libs/gst/audio/testchannels.c:
9452 Fix include of config.h
9454 2008-10-06 16:36:20 +0000 Tero Saarni <tero.saarni@gmail.com>
9456 gst-libs/gst/sdp/gstsdpmessage.c: Fix parsing of the c= field containing multicast addresses.
9457 Original commit message from CVS:
9458 Based on Patch by: Tero Saarni <tero dot saarni at gmail dot com>
9459 * gst-libs/gst/sdp/gstsdpmessage.c: (gst_sdp_parse_line),
9460 (print_media), (gst_sdp_message_dump):
9461 Fix parsing of the c= field containing multicast addresses.
9463 Add the connection info to the session or streams.
9464 Fix parsing of the bandwidth.
9465 Add debugging for the connections and bandwidths for a media.
9466 Add debugging for the bandwidth of the session.
9468 2008-10-06 16:31:27 +0000 Wim Taymans <wim.taymans@gmail.com>
9470 gst-libs/gst/rtp/gstbasertppayload.c: Configure the next seqnum and timestamp in the state change so that they can be...
9471 Original commit message from CVS:
9472 * gst-libs/gst/rtp/gstbasertppayload.c:
9473 (gst_basertppayload_change_state):
9474 Configure the next seqnum and timestamp in the state change so that they
9475 can be queried soon after.
9477 2008-10-06 16:29:33 +0000 Wim Taymans <wim.taymans@gmail.com>
9479 gst-libs/gst/rtp/gstbasertpdepayload.c: Improve debugging of the rtptime.
9480 Original commit message from CVS:
9481 * gst-libs/gst/rtp/gstbasertpdepayload.c:
9482 (gst_base_rtp_depayload_chain):
9483 Improve debugging of the rtptime.
9485 2008-10-05 11:33:47 +0000 Jan Schmidt <thaytan@mad.scientist.com>
9487 configure.ac: Back to development -> 0.10.21.1
9488 Original commit message from CVS:
9490 Back to development -> 0.10.21.1
9492 2008-10-05 08:18:31 +0000 Sebastian Dröge <slomo@circular-chaos.org>
9496 Original commit message from CVS:
9499 2008-10-05 08:11:53 +0000 Sebastian Dröge <slomo@circular-chaos.org>
9501 gst/typefind/gsttypefindfunctions.c: Add typefinder for MXF.
9502 Original commit message from CVS:
9503 * gst/typefind/gsttypefindfunctions.c: (mxf_type_find),
9505 Add typefinder for MXF.
9507 2008-10-05 08:10:09 +0000 Sebastian Dröge <slomo@circular-chaos.org>
9509 gst/typefind/gsttypefindfunctions.c: Add typefinder for MXF.
9510 Original commit message from CVS:
9511 * gst/typefind/gsttypefindfunctions.c: (mxf_type_find),
9513 Add typefinder for MXF.
9515 2008-10-03 15:19:40 +0000 Jan Schmidt <thaytan@mad.scientist.com>
9517 tests/icles/Makefile.am: Only build test-colorkey if GTK+ is available.
9518 Original commit message from CVS:
9519 * tests/icles/Makefile.am:
9520 Only build test-colorkey if GTK+ is available.
9522 === release 0.10.21 ===
9524 2008-10-03 00:03:05 +0000 Jan Schmidt <thaytan@mad.scientist.com>
9530 * docs/plugins/gst-plugins-base-plugins.args:
9531 * docs/plugins/gst-plugins-base-plugins.hierarchy:
9532 * docs/plugins/gst-plugins-base-plugins.interfaces:
9533 * docs/plugins/gst-plugins-base-plugins.prerequisites:
9534 * docs/plugins/inspect/plugin-adder.xml:
9535 * docs/plugins/inspect/plugin-alsa.xml:
9536 * docs/plugins/inspect/plugin-audioconvert.xml:
9537 * docs/plugins/inspect/plugin-audiorate.xml:
9538 * docs/plugins/inspect/plugin-audioresample.xml:
9539 * docs/plugins/inspect/plugin-audiotestsrc.xml:
9540 * docs/plugins/inspect/plugin-cdparanoia.xml:
9541 * docs/plugins/inspect/plugin-decodebin.xml:
9542 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
9543 * docs/plugins/inspect/plugin-gdp.xml:
9544 * docs/plugins/inspect/plugin-gio.xml:
9545 * docs/plugins/inspect/plugin-gnomevfs.xml:
9546 * docs/plugins/inspect/plugin-libvisual.xml:
9547 * docs/plugins/inspect/plugin-ogg.xml:
9548 * docs/plugins/inspect/plugin-pango.xml:
9549 * docs/plugins/inspect/plugin-playback.xml:
9550 * docs/plugins/inspect/plugin-queue2.xml:
9551 * docs/plugins/inspect/plugin-subparse.xml:
9552 * docs/plugins/inspect/plugin-tcp.xml:
9553 * docs/plugins/inspect/plugin-theora.xml:
9554 * docs/plugins/inspect/plugin-typefindfunctions.xml:
9555 * docs/plugins/inspect/plugin-uridecodebin.xml:
9556 * docs/plugins/inspect/plugin-video4linux.xml:
9557 * docs/plugins/inspect/plugin-videorate.xml:
9558 * docs/plugins/inspect/plugin-videoscale.xml:
9559 * docs/plugins/inspect/plugin-videotestsrc.xml:
9560 * docs/plugins/inspect/plugin-volume.xml:
9561 * docs/plugins/inspect/plugin-vorbis.xml:
9562 * docs/plugins/inspect/plugin-ximagesink.xml:
9563 * docs/plugins/inspect/plugin-xvimagesink.xml:
9564 * gst-plugins-base.doap:
9565 * win32/common/config.h:
9567 Original commit message from CVS:
9570 2008-10-02 23:44:45 +0000 Jan Schmidt <thaytan@mad.scientist.com>
9601 Original commit message from CVS:
9604 2008-09-28 22:58:18 +0000 Jan Schmidt <thaytan@mad.scientist.com>
9606 configure.ac: 0.10.20.4 pre-release
9607 Original commit message from CVS:
9609 0.10.20.4 pre-release
9611 2008-09-25 10:46:00 +0000 ogg.k.ogg.k <ogg.k.ogg.k@googlemail.com>
9613 ext/theora/theoraparse.c: Set the BOS flag on the BOS packet. Fixes #553244.
9614 Original commit message from CVS:
9615 Patch by: ogg.k.ogg.k <ogg dot k dot ogg dot k at googlemail dot com>
9616 * ext/theora/theoraparse.c: (theora_parse_set_streamheader):
9617 Set the BOS flag on the BOS packet. Fixes #553244.
9619 2008-09-23 17:48:14 +0000 Wim Taymans <wim.taymans@gmail.com>
9621 gst-libs/gst/rtsp/gstrtspmessage.c: Fix the g_return_val_if_fail() statements.
9622 Original commit message from CVS:
9623 * gst-libs/gst/rtsp/gstrtspmessage.c:
9624 (gst_rtsp_message_parse_request),
9625 (gst_rtsp_message_parse_response):
9626 Fix the g_return_val_if_fail() statements.
9628 2008-09-22 17:44:14 +0000 Michael Smith <msmith@xiph.org>
9630 gst-libs/gst/tag/gsttagdemux.c: Fail to activate if there's insufficient data in the file to be usable, preventing an...
9631 Original commit message from CVS:
9632 * gst-libs/gst/tag/gsttagdemux.c:
9633 Fail to activate if there's insufficient data in the file to be usable,
9634 preventing an assertion fail later. Fixes #552960
9636 2008-09-16 15:36:56 +0000 Jan Schmidt <thaytan@mad.scientist.com>
9638 Commit stuff that should have gone in last week when I made the pre-releases:
9639 Original commit message from CVS:
9640 Commit stuff that should have gone in last week when I made the pre-releases:
9641 2008-09-10 Jan Schmidt <jan.schmidt@sun.com>
9643 0.10.20.2 pre-release
9649 2008-09-15 15:11:18 +0000 Tim-Philipp Müller <tim@centricular.net>
9651 gst/: Recognise Kate subtitle streams (#550582).
9652 Original commit message from CVS:
9653 * gst-libs/gst/pbutils/descriptions.c:
9654 * gst/typefind/gsttypefindfunctions.c:
9655 Recognise Kate subtitle streams (#550582).
9657 2008-09-13 11:04:02 +0000 Tim-Philipp Müller <tim@centricular.net>
9659 gst-libs/gst/audio/audio.h: Remove trailing comma from enum list, which causes problems with -pendantic (#550729).
9660 Original commit message from CVS:
9661 * gst-libs/gst/audio/audio.h: (GST_AUDIO_FIELD_SIGNED):
9662 Remove trailing comma from enum list, which causes problems
9663 with -pendantic (#550729).
9665 2008-09-05 19:04:47 +0000 Tim-Philipp Müller <tim@centricular.net>
9667 gst-libs/gst/interfaces/propertyprobe.c: More sanity checks for our second-favourite interface.
9668 Original commit message from CVS:
9669 * gst-libs/gst/interfaces/propertyprobe.c:
9670 (gst_property_probe_get_properties),
9671 (gst_property_probe_get_property),
9672 (gst_property_probe_probe_property),
9673 (gst_property_probe_probe_property_name),
9674 (gst_property_probe_needs_probe),
9675 (gst_property_probe_needs_probe_name),
9676 (gst_property_probe_get_values),
9677 (gst_property_probe_get_values_name),
9678 (gst_property_probe_probe_and_get_values),
9679 (gst_property_probe_probe_and_get_values_name):
9680 More sanity checks for our second-favourite interface.
9682 2008-09-05 14:12:01 +0000 Stefan Kost <ensonic@users.sourceforge.net>
9684 gst-libs/gst/interfaces/propertyprobe.c: Check for NULL pointer, in the hope that this fixes #532864.
9685 Original commit message from CVS:
9686 * gst-libs/gst/interfaces/propertyprobe.c:
9687 Check for NULL pointer, in the hope that this fixes #532864.
9689 2008-09-05 10:24:05 +0000 Tim-Philipp Müller <tim@centricular.net>
9691 sys/xvimage/xvimagesink.c: No really, the next release is 0.10.21 (fix Since: tags in docs).
9692 Original commit message from CVS:
9693 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_class_init):
9694 No really, the next release is 0.10.21 (fix Since: tags in docs).
9696 2008-09-04 16:25:06 +0000 Wim Taymans <wim.taymans@gmail.com>
9698 gst-libs/gst/audio/gstaudiosrc.c: Disable a code path that is now called but causes a deadlock for some reason and is...
9699 Original commit message from CVS:
9700 * gst-libs/gst/audio/gstaudiosrc.c: (gst_audioringbuffer_stop):
9701 Disable a code path that is now called but causes a deadlock for some
9702 reason and is unneeded.
9704 2008-09-04 13:46:52 +0000 Stefan Kost <ensonic@users.sourceforge.net>
9706 sys/xvimage/xvimagesink.*: Add a "draw-border" property that can be set to false to disable drawing borders.
9707 Original commit message from CVS:
9708 * sys/xvimage/xvimagesink.c:
9709 * sys/xvimage/xvimagesink.h:
9710 Add a "draw-border" property that can be set to false to disable
9712 * tests/icles/test-colorkey.c:
9713 * tests/icles/Makefile.am:
9714 Add new test application for the colorkey handling.
9716 2008-09-03 14:00:06 +0000 Edward Hervey <bilboed@bilboed.com>
9718 gst-libs/gst/riff/riff-media.c: Use a decent caps for TrueSpeech instead of a ffmpeg-specific one.
9719 Original commit message from CVS:
9720 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
9721 Use a decent caps for TrueSpeech instead of a ffmpeg-specific one.
9722 This will also be fixed for upcoming gst-ffmpeg release so that once
9723 this release of -base is out, it will work with the latest gst-ffmpeg
9726 2008-09-03 13:27:20 +0000 Edward Hervey <bilboed@bilboed.com>
9728 gst-libs/gst/riff/riff-media.c: Add Truespeech mapping for RIFF formats (AVI/WAV).
9729 Original commit message from CVS:
9730 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps),
9731 (gst_riff_create_audio_template_caps):
9732 Add Truespeech mapping for RIFF formats (AVI/WAV).
9735 2008-09-03 12:23:44 +0000 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
9737 gst/typefind/gsttypefindfunctions.c: Typefind video/mj2 and image/jp2 ISO JPEG2000 mime types.
9738 Original commit message from CVS:
9739 * gst/typefind/gsttypefindfunctions.c: (plugin_init):
9740 Typefind video/mj2 and image/jp2 ISO JPEG2000 mime types.
9743 2008-09-03 10:12:04 +0000 Stefan Kost <ensonic@users.sourceforge.net>
9745 Rework last change, so that we build subparse, but just disable the sami parse functionality, if we're configured to ...
9746 Original commit message from CVS:
9748 * gst/subparse/Makefile.am:
9749 * gst/subparse/gstsubparse.c:
9750 * gst/subparse/samiparse.c:
9751 * tests/check/elements/subparse.c:
9752 Rework last change, so that we build subparse, but just disable the
9753 sami parse functionality, if we're configured to not use xml. In the
9754 tests only the sami test is disabled now.
9756 2008-09-02 15:07:09 +0000 Stefan Kost <ensonic@users.sourceforge.net>
9758 configure.ac: Disable subparse when xml is disabled. It woundn't work anyway. Fixes test runs.
9759 Original commit message from CVS:
9761 Disable subparse when xml is disabled. It woundn't work anyway. Fixes
9764 2008-09-02 09:33:17 +0000 Tim-Philipp Müller <tim@centricular.net>
9766 po/POTFILES.in: Add some more files with strings for translation.
9767 Original commit message from CVS:
9769 Add some more files with strings for translation.
9771 2008-09-02 06:37:04 +0000 Stefan Kost <ensonic@users.sourceforge.net>
9773 Use new geo location tags from core. Fixes #481169
9774 Original commit message from CVS:
9775 * gst-libs/gst/tag/gstvorbistag.c:
9776 * tests/check/libs/tag.c:
9777 Use new geo location tags from core. Fixes #481169
9779 2008-09-01 16:05:45 +0000 Edward Hervey <bilboed@bilboed.com>
9781 tests/check/elements/audioresample.c: Now that GstBaseTransform is 'fixed' ... remove cruft from tests.
9782 Original commit message from CVS:
9783 * tests/check/elements/audioresample.c: (setup_audioresample),
9784 (fail_unless_perfect_stream), (test_perfect_stream_instance),
9785 (test_discont_stream_instance):
9786 Now that GstBaseTransform is 'fixed' ... remove cruft from tests.
9787 Add debugging for coherence.
9789 2008-08-30 15:55:06 +0000 Jonathan Matthew <notverysmart@gmail.com>
9791 gst/typefind/gsttypefindfunctions.c: Add typefinder for PDF documents (which is nice to have, since it's a common for...
9792 Original commit message from CVS:
9793 Patch by: Jonathan Matthew <notverysmart gmail com>
9794 * gst/typefind/gsttypefindfunctions.c: (plugin_init):
9795 Add typefinder for PDF documents (which is nice to have, since it's a
9796 common format, but also helps prevent false positives). Fixes #549814.
9798 2008-08-27 15:30:16 +0000 Wim Taymans <wim.taymans@gmail.com>
9800 gst/playback/gstplaybin2.c: Fix nasty race where multiple decodebins could start pushing data before we manage to con...
9801 Original commit message from CVS:
9802 * gst/playback/gstplaybin2.c: (selector_blocked), (pad_added_cb),
9804 Fix nasty race where multiple decodebins could start pushing data before
9805 we manage to configure the sinks, resulting in not-linked errors in
9806 typical RTSP streaming cases.
9808 2008-08-26 17:24:31 +0000 Wim Taymans <wim.taymans@gmail.com>
9810 gst-libs/gst/audio/gstaudiosink.c: Since we now call stop, we trigger this code path that causes a deadlock is appare...
9811 Original commit message from CVS:
9812 * gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_stop):
9813 Since we now call stop, we trigger this code path that causes a deadlock
9814 is apparently not needed.
9816 2008-08-26 15:45:36 +0000 Wim Taymans <wim.taymans@gmail.com>
9818 gst-libs/gst/audio/gstringbuffer.c: Also allow the case where the ringbuffer was paused when we try to stop it so tha...
9819 Original commit message from CVS:
9820 * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_start),
9821 (gst_ring_buffer_stop):
9822 Also allow the case where the ringbuffer was paused when we try to stop
9823 it so that the basesrc stop function is still called.
9825 2008-08-23 15:25:44 +0000 Mike Ruprecht <cmaiku@gmail.com>
9827 sys/v4l/gstv4lelement.c: Reprobe devices again instead of taking a cached list as new devices could've been plugged i...
9828 Original commit message from CVS:
9829 Patch by: Mike Ruprecht <cmaiku at gmail dot com>
9830 * sys/v4l/gstv4lelement.c: (gst_v4l_class_probe_devices):
9831 Reprobe devices again instead of taking a cached list as new
9832 devices could've been plugged in. Fixes bug #549062.
9834 2008-08-23 15:19:59 +0000 Alessandro Dessina <alessandro@nnva.org>
9836 ext/ogg/gstoggdemux.c: Don't add pads and activate them for skeleton streams. These are already handled inside oggdem...
9837 Original commit message from CVS:
9838 Patch by: Alessandro Dessina <alessandro nnva org>
9839 * ext/ogg/gstoggdemux.c: (gst_ogg_demux_deactivate_current_chain),
9840 (gst_ogg_demux_activate_chain):
9841 Don't add pads and activate them for skeleton streams. These are already
9842 handled inside oggdemux. Fixes bug #537599.
9844 2008-08-22 15:54:15 +0000 Wim Taymans <wim.taymans@gmail.com>
9846 ext/vorbis/vorbisdec.c: Reset variable so that query and convert fail after going back to
9847 Original commit message from CVS:
9848 * ext/vorbis/vorbisdec.c: (vorbis_dec_change_state):
9849 Reset variable so that query and convert fail after going back to
9850 READY. Fixes #548898.
9852 2008-08-22 07:24:13 +0000 Sebastian Dröge <slomo@circular-chaos.org>
9854 ext/vorbis/vorbisenc.c: If a buffer arrives with a timestamp before the timestamp+duration of the previous buffer cli...
9855 Original commit message from CVS:
9856 * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_chain):
9857 If a buffer arrives with a timestamp before the timestamp+duration
9858 of the previous buffer clip it instead of dropping it completely.
9859 Slight improvement for the unfixable bug #548913.
9861 2008-08-21 14:19:21 +0000 Sebastian Dröge <slomo@circular-chaos.org>
9863 ext/vorbis/vorbisdec.c: Take the current timestamp instead of timestamp+duration for the offset.
9864 Original commit message from CVS:
9865 * ext/vorbis/vorbisdec.c: (vorbis_handle_data_packet):
9866 Take the current timestamp instead of timestamp+duration for the offset.
9867 This offset will later be used for calculating the timestamp and
9868 otherwise vorbisdec will interpolate timestamps wrong if upstream
9869 only sends timestamps and no granulepos.
9871 2008-08-21 11:20:36 +0000 Stefan Kost <ensonic@users.sourceforge.net>
9873 tests/examples/seek/seek.c: Don't crash when having no visualisations.
9874 Original commit message from CVS:
9875 * tests/examples/seek/seek.c:
9876 Don't crash when having no visualisations.
9878 2008-08-16 20:57:27 +0000 David Schleef <ds@schleef.org>
9880 gst/typefind/gsttypefindfunctions.c: DV typefinding. Remove check for a bit that is 0 in IEC 61384, but not SMPTE 314M.
9881 Original commit message from CVS:
9882 * gst/typefind/gsttypefindfunctions.c: DV typefinding. Remove
9883 check for a bit that is 0 in IEC 61384, but not SMPTE 314M.
9886 2008-08-15 07:24:38 +0000 Sebastian Dröge <slomo@circular-chaos.org>
9888 gst-libs/gst/pbutils/missing-plugins.c: When cleaning up the caps fields also remove "depth" for the same reason we r...
9889 Original commit message from CVS:
9890 * gst-libs/gst/pbutils/missing-plugins.c: (copy_and_clean_caps):
9891 When cleaning up the caps fields also remove "depth" for the same
9892 reason we remove "width".
9894 2008-08-14 17:14:53 +0000 Tim-Philipp Müller <tim@centricular.net>
9896 gst-libs/gst/pbutils/descriptions.c: Add Lead H.264 here as well.
9897 Original commit message from CVS:
9898 * gst-libs/gst/pbutils/descriptions.c: (format_info_get_desc):
9899 Add Lead H.264 here as well.
9901 2008-08-14 15:17:31 +0000 Julien Moutte <julien@moutte.net>
9903 gst-libs/gst/riff/riff-media.c: Add Lead H.264 variant.
9904 Original commit message from CVS:
9905 2008-08-14 Julien Moutte <julien@fluendo.com>
9906 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
9907 (gst_riff_create_video_template_caps): Add Lead H.264 variant.
9909 2008-08-13 09:17:38 +0000 Wim Taymans <wim.taymans@gmail.com>
9911 gst-libs/gst/audio/gstbaseaudiosrc.c: When not slaved to another clock also subtract the base_time from our internal ...
9912 Original commit message from CVS:
9913 * gst-libs/gst/audio/gstbaseaudiosrc.c:
9914 (gst_base_audio_src_create):
9915 When not slaved to another clock also subtract the base_time from our
9916 internal clock time to get the running time.
9918 2008-08-13 00:59:07 +0000 David Schleef <ds@schleef.org>
9920 ext/theora/theoraenc.c: Remove the 2000 kbit limit to bitrate, since it has no basis in libtheora.
9921 Original commit message from CVS:
9922 * ext/theora/theoraenc.c: Remove the 2000 kbit limit to bitrate,
9923 since it has no basis in libtheora.
9925 2008-08-12 06:31:49 +0000 Stefan Kost <ensonic@users.sourceforge.net>
9927 gst-libs/gst/interfaces/propertyprobe.h: Remove double "interface" from doc-string.
9928 Original commit message from CVS:
9929 * gst-libs/gst/interfaces/propertyprobe.h:
9930 Remove double "interface" from doc-string.
9931 * gst-libs/gst/interfaces/xoverlay.h:
9933 * gst-libs/gst/riff/riff.c:
9934 Add basic doc blobs.
9936 2008-08-11 15:05:35 +0000 Stefan Kost <ensonic@users.sourceforge.net>
9938 gst-libs/gst/audio/Makefile.am: Don't try to build that example anymore.
9939 Original commit message from CVS:
9940 * gst-libs/gst/audio/Makefile.am:
9941 Don't try to build that example anymore.
9943 2008-08-11 14:51:58 +0000 Stefan Kost <ensonic@users.sourceforge.net>
9945 gst-libs/gst/audio/: Move audiofiltertemplate to gst-template.
9946 Original commit message from CVS:
9947 * gst-libs/gst/audio/.cvsignore:
9948 * gst-libs/gst/audio/Makefile.am:
9949 * gst-libs/gst/audio/gstaudiofiltertemplate.c:
9950 * gst-libs/gst/audio/make_filter:
9951 Move audiofiltertemplate to gst-template.
9953 2008-08-11 09:20:33 +0000 Stefan Kost <ensonic@users.sourceforge.net>
9955 More docs and shuffling. What can we do with the hundreds of #defines.
9956 Original commit message from CVS:
9957 * docs/libs/gst-plugins-base-libs-sections.txt:
9958 * gst-libs/gst/audio/gstaudiosrc.h:
9959 More docs and shuffling. What can we do with the hundreds of #defines.
9961 2008-08-11 08:34:56 +0000 Stefan Kost <ensonic@users.sourceforge.net>
9963 gst-libs/gst/: Reducing number of dundocumented symbols.
9964 Original commit message from CVS:
9965 * gst-libs/gst/audio/audio.h:
9966 * gst-libs/gst/audio/gstaudiofilter.h:
9967 * gst-libs/gst/audio/gstringbuffer.h:
9968 * gst-libs/gst/interfaces/propertyprobe.h:
9969 * gst-libs/gst/tag/gsttagdemux.h:
9970 Reducing number of dundocumented symbols.
9972 2008-08-11 07:16:30 +0000 Stefan Kost <ensonic@users.sourceforge.net>
9974 gst-libs/gst/audio/audio.c: Fix doc comment syntax.
9975 Original commit message from CVS:
9976 * gst-libs/gst/audio/audio.c:
9977 Fix doc comment syntax.
9978 * gst-libs/gst/interfaces/propertyprobe.c:
9979 Add more doc-comments and a FIXME: for the signal.
9981 2008-08-07 16:11:14 +0000 Sebastian Dröge <slomo@circular-chaos.org>
9983 ext/ogg/gstoggmux.*: Don't pretend to support NEWSEGMENT events, instead override the
9984 Original commit message from CVS:
9985 * ext/ogg/gstoggmux.c: (gst_ogg_mux_sink_event),
9986 (gst_ogg_mux_request_new_pad):
9987 * ext/ogg/gstoggmux.h:
9988 Don't pretend to support NEWSEGMENT events, instead override the
9989 GstCollectPads event function to return FALSE on NEWSEGMENT events
9990 and do the normal work for other events.
9991 This prevents elements like flacenc to seek to the start and rewrite
9992 some data which then results in a broken Ogg packet.
9994 2008-08-07 15:58:58 +0000 Frederic Crozat <fcrozat@mandriva.org>
9996 Make sure gettext returns translations in UTF-8 encoding rather than in the current locale encoding (#546822).
9997 Original commit message from CVS:
9998 Patch by: Frederic Crozat <fcrozat@mandriva.org>
9999 * ext/alsa/gstalsaplugin.c: (plugin_init):
10000 * ext/cdparanoia/gstcdparanoiasrc.c: (plugin_init):
10001 * ext/gnomevfs/gstgnomevfs.c: (plugin_init):
10002 * ext/ogg/gstoggdemux.c: (gst_ogg_demux_plugin_init):
10003 * gst-libs/gst/audio/gstbaseaudiosrc.c: (_do_init):
10004 * gst-libs/gst/pbutils/pbutils.c: (gst_pb_utils_init):
10005 * gst-libs/gst/tag/tags.c: (gst_tag_register_tags_internal):
10006 * gst/playback/gstdecodebin.c: (plugin_init):
10007 * gst/playback/gstdecodebin2.c: (gst_decode_bin_plugin_init):
10008 * gst/playback/gstplayback.c: (plugin_init):
10009 * gst/playback/gstqueue2.c: (plugin_init):
10010 * gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_plugin_init):
10011 * sys/v4l/gstv4l.c: (plugin_init):
10012 Make sure gettext returns translations in UTF-8 encoding rather
10013 than in the current locale encoding (#546822).
10015 2008-08-06 13:12:07 +0000 Stefan Kost <ensonic@users.sourceforge.net>
10017 gst-libs/gst/pbutils/descriptions.c: Add audio/x-qdm for qtdemux.
10018 Original commit message from CVS:
10019 * gst-libs/gst/pbutils/descriptions.c:
10020 Add audio/x-qdm for qtdemux.
10022 2008-08-05 15:38:06 +0000 Stefan Kost <ensonic@users.sourceforge.net>
10024 ext/vorbis/vorbisdec.c: Do not leak old taglist.
10025 Original commit message from CVS:
10026 * ext/vorbis/vorbisdec.c:
10027 Do not leak old taglist.
10029 2008-08-04 12:35:07 +0000 Stefan Kost <ensonic@users.sourceforge.net>
10031 tests/icles/test-scale.c: Include <stdlib.h> for atoi().
10032 Original commit message from CVS:
10033 * tests/icles/test-scale.c:
10034 Include <stdlib.h> for atoi().
10036 2008-08-04 09:11:08 +0000 Andy Wingo <wingo@pobox.com>
10038 gst/audiotestsrc/gstaudiotestsrc.c: Very crucial and important documentation fix.
10039 Original commit message from CVS:
10040 2008-08-04 Andy Wingo <wingo@pobox.com>
10041 * gst/audiotestsrc/gstaudiotestsrc.c: Very crucial and important
10044 2008-08-01 13:06:59 +0000 Stefan Kost <ensonic@users.sourceforge.net>
10046 gst/adder/gstadder.c: Cleanup lots of empty lines that came from gst-indent going havoc before I added the INDENT_ON/...
10047 Original commit message from CVS:
10048 * gst/adder/gstadder.c:
10049 Cleanup lots of empty lines that came from gst-indent going havoc
10050 before I added the INDENT_ON/OFF marker some time agao.
10052 2008-08-01 11:55:07 +0000 Stefan Kost <ensonic@users.sourceforge.net>
10054 Bump requirement to latest core and use new tag for riff formats.
10055 Original commit message from CVS:
10057 * gst-libs/gst/riff/riff-read.c:
10058 Bump requirement to latest core and use new tag for riff formats.
10059 Needed for #520694.
10061 2008-08-01 11:14:49 +0000 Wim Taymans <wim.taymans@gmail.com>
10063 tests/examples/dynamic/: Add example app that dynamically switches between 3 'encoders'.
10064 Original commit message from CVS:
10065 * tests/examples/dynamic/Makefile.am:
10066 * tests/examples/dynamic/codec-select.c: (make_encoder),
10067 (make_pipeline), (do_switch), (my_bus_callback), (main):
10068 Add example app that dynamically switches between 3 'encoders'.
10070 2008-07-31 13:06:13 +0000 Wim Taymans <wim.taymans@gmail.com>
10072 gst/playback/gstplaysink.c: Add some more comments.
10073 Original commit message from CVS:
10074 * gst/playback/gstplaysink.c: (gst_play_sink_set_vis_plugin):
10075 Add some more comments.
10077 2008-07-31 12:58:44 +0000 Wim Taymans <wim.taymans@gmail.com>
10079 gst/videotestsrc/gstvideotestsrc.c: Discard buffers of the wrong size after renegotiation, this is perfectly possible...
10080 Original commit message from CVS:
10081 * gst/videotestsrc/gstvideotestsrc.c: (gst_video_test_src_getcaps),
10082 (gst_video_test_src_create):
10083 Discard buffers of the wrong size after renegotiation, this is perfectly
10084 possible with things like capsfilter that could suggest caps changes
10085 upstream without knowing the size of the buffer.
10087 2008-07-31 11:39:44 +0000 Wim Taymans <wim.taymans@gmail.com>
10089 tests/icles/: Add dynamic rescaling tests for the new basetransform.
10090 Original commit message from CVS:
10091 * tests/icles/.cvsignore:
10092 * tests/icles/Makefile.am:
10093 * tests/icles/test-scale.c: (make_pipeline), (main):
10094 Add dynamic rescaling tests for the new basetransform.
10096 2008-07-30 19:51:36 +0000 Tim-Philipp Müller <tim@centricular.net>
10098 gst/audioconvert/Makefile.am: Dist recently-added gstfastrandom.h.
10099 Original commit message from CVS:
10100 * gst/audioconvert/Makefile.am:
10101 Dist recently-added gstfastrandom.h.
10103 2008-07-30 15:29:44 +0000 Edward Hervey <bilboed@bilboed.com>
10105 sys/xvimage/xvimagesink.c: Fix a "may be used uninitialized in this function" which weirdly only appears on macosx (?).
10106 Original commit message from CVS:
10107 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support):
10108 Fix a "may be used uninitialized in this function" which weirdly only
10109 appears on macosx (?).
10111 2008-07-30 09:02:31 +0000 Stefan Kost <ensonic@users.sourceforge.net>
10113 gst-libs/gst/riff/riff-ids.h: Adding acid chunk for tempo and loop information.
10114 Original commit message from CVS:
10115 * gst-libs/gst/riff/riff-ids.h:
10116 Adding acid chunk for tempo and loop information.
10118 2008-07-29 13:01:13 +0000 Stefan Kost <ensonic@users.sourceforge.net>
10120 sys/xvimage/Makefile.am: floor() needs linking to $(LIBM).
10121 Original commit message from CVS:
10122 * sys/xvimage/Makefile.am:
10123 floor() needs linking to $(LIBM).
10125 2008-07-29 12:35:54 +0000 Stefan Kost <ensonic@users.sourceforge.net>
10127 ext/gnomevfs/gstgnomevfssrc.c: Aggregate short reads and add some comments and debug logging.
10128 Original commit message from CVS:
10129 * ext/gnomevfs/gstgnomevfssrc.c:
10130 Aggregate short reads and add some comments and debug logging.
10133 2008-07-29 10:26:28 +0000 Stefan Kost <ensonic@users.sourceforge.net>
10135 gst/playback/gstplaybasebin.c: Fix property doc markup (its not a signal).
10136 Original commit message from CVS:
10137 * gst/playback/gstplaybasebin.c:
10138 Fix property doc markup (its not a signal).
10139 * sys/xvimage/xvimagesink.c:
10140 Add since tag for new proeprties (also add sice tags fro the last two
10143 2008-07-29 08:59:32 +0000 Stefan Kost <ensonic@users.sourceforge.net>
10145 sys/xvimage/xvimagesink.*: Add autofill/colorkey properties. Fixes #538656.
10146 Original commit message from CVS:
10147 * sys/xvimage/xvimagesink.c:
10148 * sys/xvimage/xvimagesink.h:
10149 Add autofill/colorkey properties. Fixes #538656.
10151 2008-07-29 01:58:05 +0000 David Schleef <ds@schleef.org>
10153 sys/xvimage/xvimagesink.c: Fix rounding errors when converting colorbalance values between hardware and object proper...
10154 Original commit message from CVS:
10155 * sys/xvimage/xvimagesink.c:
10156 Fix rounding errors when converting colorbalance values
10157 between hardware and object property ranges. Partial
10158 fix for #537889, however, there still seems to be a small
10159 drift problem that could be totem's fault.
10161 2008-07-28 15:34:13 +0000 Sebastian Dröge <slomo@circular-chaos.org>
10163 ext/ogg/gstoggdemux.c: Don't use GST_CLOCK_TIME_NONE as start of NEWSEGMENT events.
10164 Original commit message from CVS:
10165 * ext/ogg/gstoggdemux.c: (gst_ogg_demux_chain_peer),
10166 (gst_ogg_demux_perform_seek), (gst_ogg_demux_handle_page):
10167 Don't use GST_CLOCK_TIME_NONE as start of NEWSEGMENT events.
10168 This fixes a critical warning.
10170 2008-07-28 13:12:51 +0000 Sebastian Dröge <slomo@circular-chaos.org>
10172 ext/ogg/gstoggmux.c: Allow muxing of CELT into Ogg streams.
10173 Original commit message from CVS:
10174 * ext/ogg/gstoggmux.c:
10175 Allow muxing of CELT into Ogg streams.
10177 2008-07-28 12:47:06 +0000 Sebastian Dröge <slomo@circular-chaos.org>
10179 gst/typefind/gsttypefindfunctions.c: Add simple typefinder for the CELT codec (www.celt-codec.org).
10180 Original commit message from CVS:
10181 * gst/typefind/gsttypefindfunctions.c: (celt_type_find),
10183 Add simple typefinder for the CELT codec (www.celt-codec.org).
10185 2008-07-27 11:12:41 +0000 Jan Gerber <j@oil21.org>
10187 ext/ogg/gstoggdemux.c: Fix calculation of the start time from skeleton streams.
10188 Original commit message from CVS:
10189 Patch by: Jan Gerber <j at oil21 dot org>
10190 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_parse_skeleton_fisbone):
10191 Fix calculation of the start time from skeleton streams.
10194 2008-07-24 13:19:26 +0000 Stefan Kost <ensonic@users.sourceforge.net>
10196 tests/examples/seek/seek.c: Use 64 bit constant GST_CLOCK_TIME_NONE instead of plain -1.
10197 Original commit message from CVS:
10198 * tests/examples/seek/seek.c:
10199 Use 64 bit constant GST_CLOCK_TIME_NONE instead of plain -1.
10201 2008-07-23 18:34:19 +0000 Sebastian Dröge <slomo@circular-chaos.org>
10203 gst/audioconvert/: Implement a linear congruential generator as pseudo random number generator for the dither noise. ...
10204 Original commit message from CVS:
10205 * gst/audioconvert/audioconvert.h:
10206 * gst/audioconvert/gstaudioquantize.c:
10207 (gst_audio_quantize_setup_dither),
10208 (gst_audio_quantize_free_dither):
10209 * gst/audioconvert/gstfastrandom.h:
10210 Implement a linear congruential generator as pseudo random number
10211 generator for the dither noise. This is about 2 times faster than
10212 using GLib's mersenne twister. Also this uses only integer math for
10213 generating integers while GLib internally uses floating point math.
10215 2008-07-23 18:27:15 +0000 Michael Smith <msmith@xiph.org>
10217 configure.ac: Remove AC_ISC_POSIX; it breaks on some systems and is not needed.
10218 Original commit message from CVS:
10220 Remove AC_ISC_POSIX; it breaks on some systems and is not needed.
10222 2008-07-23 13:17:31 +0000 Damien Lespiau <damien.lespiau@gmail.com>
10224 gst-libs/gst/sdp/gstsdpmessage.c: Use GST_STR_NULL to avoid crashes with libcs that don't like NULL strings in printf...
10225 Original commit message from CVS:
10226 Patch by: Damien Lespiau <damien.lespiau gmail com>
10227 * gst-libs/gst/sdp/gstsdpmessage.c: (print_media):
10228 Use GST_STR_NULL to avoid crashes with libcs that don't
10229 like NULL strings in printf args (such as the win32 one).
10232 2008-07-17 14:21:30 +0000 Jan Schmidt <thaytan@mad.scientist.com>
10234 sys/xvimage/xvimagesink.c: Oops - set the size of the image used for probing back to 1x1, for consistency with ximage...
10235 Original commit message from CVS:
10236 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_check_xshm_calls):
10237 Oops - set the size of the image used for probing back to 1x1, for
10238 consistency with ximagesink
10240 2008-07-17 13:57:33 +0000 Jan Schmidt <thaytan@mad.scientist.com>
10242 sys/: it's not legal to ask the
10243 Original commit message from CVS:
10244 * sys/ximage/ximagesink.c: (gst_ximagesink_check_xshm_calls),
10245 (gst_ximagesink_ximage_new):
10246 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_check_xshm_calls),
10247 (gst_xvimagesink_xvimage_new):
10248 Apparently on Solaris and OS/X (at least), it's not legal to ask the
10249 X server to attach to a shared memory segment after we've deleted it,
10250 with the result that MIT-SHM is disabled. Instead, remove it only after
10251 X succeeds in attaching too.
10253 2008-07-17 02:30:24 +0000 David Schleef <ds@schleef.org>
10255 gst/audiotestsrc/gstaudiotestsrc.*: Add 'ticks', a 1/30 second sine wave pulse every second.
10256 Original commit message from CVS:
10257 * gst/audiotestsrc/gstaudiotestsrc.c:
10258 * gst/audiotestsrc/gstaudiotestsrc.h:
10259 Add 'ticks', a 1/30 second sine wave pulse every second.
10261 2008-07-15 22:43:16 +0000 David Schleef <ds@schleef.org>
10263 gst-libs/gst/video/video.c: Revert ABI change.
10264 Original commit message from CVS:
10265 * gst-libs/gst/video/video.c: Revert ABI change.
10267 2008-07-15 13:05:04 +0000 Sebastian Dröge <slomo@circular-chaos.org>
10269 gst-libs/gst/riff/riff-media.c: Make it impossible to have NULL caps at the point where we set framerate and other th...
10270 Original commit message from CVS:
10271 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
10272 Make it impossible to have NULL caps at the point where we set
10273 framerate and other things. Also don't return immediately for "3ivd"
10274 video and let framerate, etc be set. Might fix bug #542508.
10276 2008-07-14 17:06:26 +0000 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
10278 gst-libs/gst/video/video.c: Video format can also be conveniently determined from (many) non-fixed caps.
10279 Original commit message from CVS:
10280 * gst-libs/gst/video/video.c: (gst_video_format_parse_caps):
10281 Video format can also be conveniently determined from (many)
10284 2008-07-14 08:18:58 +0000 Jan Schmidt <thaytan@mad.scientist.com>
10286 gst/playback/: First stab at integrating DVD subpicture overlay into playbin. Successfully plugs and plays, but the q...
10287 Original commit message from CVS:
10288 * gst/playback/gstplaybasebin.c:
10289 * gst/playback/gstplaybasebin.h:
10290 * gst/playback/gstplaybin.c:
10291 * gst/playback/gststreamselector.c:
10292 First stab at integrating DVD subpicture overlay into
10293 playbin. Successfully plugs and plays, but the queues need
10294 shrinking - 3 seconds of video is too much buffering.
10296 2008-07-11 18:06:33 +0000 Stefan Kost <ensonic@users.sourceforge.net>
10298 gst/audioconvert/gstaudioconvert.c: Remove now obsolete note in the docs.
10299 Original commit message from CVS:
10300 * gst/audioconvert/gstaudioconvert.c:
10301 Remove now obsolete note in the docs.
10303 2008-07-11 06:10:24 +0000 Stefan Kost <ensonic@users.sourceforge.net>
10305 Cleanup Plugin docs. Link to signals and properties. Fix sub-section titles. Drop mentining that all our example pipe...
10306 Original commit message from CVS:
10307 * docs/plugins/gst-plugins-base-plugins-docs.sgml:
10308 * docs/plugins/gst-plugins-base-plugins-overrides.txt:
10309 * docs/plugins/gst-plugins-base-plugins-sections.txt:
10310 * docs/plugins/gst-plugins-base-plugins.args:
10311 * docs/plugins/gst-plugins-base-plugins.hierarchy:
10312 * docs/plugins/gst-plugins-base-plugins.interfaces:
10313 * docs/plugins/gst-plugins-base-plugins.prerequisites:
10314 * docs/plugins/gst-plugins-base-plugins.signals:
10315 * docs/plugins/inspect/plugin-adder.xml:
10316 * docs/plugins/inspect/plugin-alsa.xml:
10317 * docs/plugins/inspect/plugin-audioconvert.xml:
10318 * docs/plugins/inspect/plugin-audiorate.xml:
10319 * docs/plugins/inspect/plugin-audioresample.xml:
10320 * docs/plugins/inspect/plugin-audiotestsrc.xml:
10321 * docs/plugins/inspect/plugin-cdparanoia.xml:
10322 * docs/plugins/inspect/plugin-decodebin.xml:
10323 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
10324 * docs/plugins/inspect/plugin-gdp.xml:
10325 * docs/plugins/inspect/plugin-gnomevfs.xml:
10326 * docs/plugins/inspect/plugin-libvisual.xml:
10327 * docs/plugins/inspect/plugin-ogg.xml:
10328 * docs/plugins/inspect/plugin-pango.xml:
10329 * docs/plugins/inspect/plugin-playback.xml:
10330 * docs/plugins/inspect/plugin-queue2.xml:
10331 * docs/plugins/inspect/plugin-subparse.xml:
10332 * docs/plugins/inspect/plugin-tcp.xml:
10333 * docs/plugins/inspect/plugin-theora.xml:
10334 * docs/plugins/inspect/plugin-typefindfunctions.xml:
10335 * docs/plugins/inspect/plugin-uridecodebin.xml:
10336 * docs/plugins/inspect/plugin-video4linux.xml:
10337 * docs/plugins/inspect/plugin-videorate.xml:
10338 * docs/plugins/inspect/plugin-videoscale.xml:
10339 * docs/plugins/inspect/plugin-videotestsrc.xml:
10340 * docs/plugins/inspect/plugin-volume.xml:
10341 * docs/plugins/inspect/plugin-vorbis.xml:
10342 * docs/plugins/inspect/plugin-ximagesink.xml:
10343 * docs/plugins/inspect/plugin-xvimagesink.xml:
10344 * ext/alsa/gstalsamixer.c:
10345 * ext/alsa/gstalsasink.c:
10346 * ext/alsa/gstalsasrc.c:
10347 * ext/gio/gstgiosink.c:
10348 * ext/gio/gstgiosrc.c:
10349 * ext/gio/gstgiostreamsink.c:
10350 * ext/gio/gstgiostreamsrc.c:
10351 * ext/gnomevfs/gstgnomevfssink.c:
10352 * ext/gnomevfs/gstgnomevfssrc.c:
10353 * ext/ogg/gstoggdemux.c:
10354 * ext/ogg/gstoggmux.c:
10355 * ext/pango/gstclockoverlay.c:
10356 * ext/pango/gsttextoverlay.c:
10357 * ext/pango/gsttextrender.c:
10358 * ext/pango/gsttimeoverlay.c:
10359 * ext/theora/theoradec.c:
10360 * ext/theora/theoraenc.c:
10361 * ext/theora/theoraparse.c:
10362 * ext/vorbis/vorbisdec.c:
10363 * ext/vorbis/vorbisenc.c:
10364 * ext/vorbis/vorbisparse.c:
10365 * ext/vorbis/vorbistag.c:
10366 * gst/adder/gstadder.c:
10367 * gst/audioconvert/gstaudioconvert.c:
10368 * gst/audioresample/gstaudioresample.c:
10369 * gst/audiotestsrc/gstaudiotestsrc.c:
10370 * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
10371 * gst/gdp/gstgdpdepay.c:
10372 * gst/gdp/gstgdppay.c:
10373 * gst/playback/gstdecodebin2.c:
10374 * gst/playback/gstplaybin.c:
10375 * gst/playback/gstplaybin2.c:
10376 * gst/playback/gstqueue2.c:
10377 * gst/playback/gsturidecodebin.c:
10378 * gst/tcp/gstmultifdsink.c:
10379 * gst/tcp/gsttcpserversink.c:
10380 * gst/videorate/gstvideorate.c:
10381 * gst/videoscale/gstvideoscale.c:
10382 * gst/videotestsrc/gstvideotestsrc.c:
10383 * gst/volume/gstvolume.c:
10384 * sys/ximage/ximagesink.c:
10385 * sys/xvimage/xvimagesink.c:
10386 Cleanup Plugin docs. Link to signals and properties. Fix sub-section
10387 titles. Drop mentining that all our example pipelines are "simple"
10390 2008-07-10 21:06:06 +0000 Stefan Kost <ensonic@users.sourceforge.net>
10392 Cleanup Plugin docs. Link to signals and properties. Fix sub-section titles. Drop mentining that all our example pipe...
10393 Original commit message from CVS:
10394 * docs/plugins/gst-plugins-base-plugins-docs.sgml:
10395 * docs/plugins/gst-plugins-base-plugins-overrides.txt:
10396 * docs/plugins/gst-plugins-base-plugins-sections.txt:
10397 * docs/plugins/gst-plugins-base-plugins.args:
10398 * docs/plugins/gst-plugins-base-plugins.hierarchy:
10399 * docs/plugins/gst-plugins-base-plugins.interfaces:
10400 * docs/plugins/gst-plugins-base-plugins.prerequisites:
10401 * docs/plugins/gst-plugins-base-plugins.signals:
10402 * docs/plugins/inspect/plugin-adder.xml:
10403 * docs/plugins/inspect/plugin-alsa.xml:
10404 * docs/plugins/inspect/plugin-audioconvert.xml:
10405 * docs/plugins/inspect/plugin-audiorate.xml:
10406 * docs/plugins/inspect/plugin-audioresample.xml:
10407 * docs/plugins/inspect/plugin-audiotestsrc.xml:
10408 * docs/plugins/inspect/plugin-cdparanoia.xml:
10409 * docs/plugins/inspect/plugin-decodebin.xml:
10410 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
10411 * docs/plugins/inspect/plugin-gdp.xml:
10412 * docs/plugins/inspect/plugin-gnomevfs.xml:
10413 * docs/plugins/inspect/plugin-libvisual.xml:
10414 * docs/plugins/inspect/plugin-ogg.xml:
10415 * docs/plugins/inspect/plugin-pango.xml:
10416 * docs/plugins/inspect/plugin-playback.xml:
10417 * docs/plugins/inspect/plugin-queue2.xml:
10418 * docs/plugins/inspect/plugin-subparse.xml:
10419 * docs/plugins/inspect/plugin-tcp.xml:
10420 * docs/plugins/inspect/plugin-theora.xml:
10421 * docs/plugins/inspect/plugin-typefindfunctions.xml:
10422 * docs/plugins/inspect/plugin-uridecodebin.xml:
10423 * docs/plugins/inspect/plugin-video4linux.xml:
10424 * docs/plugins/inspect/plugin-videorate.xml:
10425 * docs/plugins/inspect/plugin-videoscale.xml:
10426 * docs/plugins/inspect/plugin-videotestsrc.xml:
10427 * docs/plugins/inspect/plugin-volume.xml:
10428 * docs/plugins/inspect/plugin-vorbis.xml:
10429 * docs/plugins/inspect/plugin-ximagesink.xml:
10430 * docs/plugins/inspect/plugin-xvimagesink.xml:
10431 * ext/alsa/gstalsamixer.c:
10432 * ext/alsa/gstalsasink.c:
10433 * ext/alsa/gstalsasrc.c:
10434 * ext/gio/gstgiosink.c:
10435 * ext/gio/gstgiosrc.c:
10436 * ext/gio/gstgiostreamsink.c:
10437 * ext/gio/gstgiostreamsrc.c:
10438 * ext/gnomevfs/gstgnomevfssink.c:
10439 * ext/gnomevfs/gstgnomevfssrc.c:
10440 * ext/ogg/gstoggdemux.c:
10441 * ext/ogg/gstoggmux.c:
10442 * ext/pango/gstclockoverlay.c:
10443 * ext/pango/gsttextoverlay.c:
10444 * ext/pango/gsttextrender.c:
10445 * ext/pango/gsttimeoverlay.c:
10446 * ext/theora/theoradec.c:
10447 * ext/theora/theoraenc.c:
10448 * ext/theora/theoraparse.c:
10449 * ext/vorbis/vorbisdec.c:
10450 * ext/vorbis/vorbisenc.c:
10451 * ext/vorbis/vorbisparse.c:
10452 * ext/vorbis/vorbistag.c:
10453 * gst/adder/gstadder.c:
10454 * gst/audioconvert/gstaudioconvert.c:
10455 * gst/audioresample/gstaudioresample.c:
10456 * gst/audiotestsrc/gstaudiotestsrc.c:
10457 * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
10458 * gst/gdp/gstgdpdepay.c:
10459 * gst/gdp/gstgdppay.c:
10460 * gst/playback/gstdecodebin2.c:
10461 * gst/playback/gstplaybin.c:
10462 * gst/playback/gstplaybin2.c:
10463 * gst/playback/gstqueue2.c:
10464 * gst/playback/gsturidecodebin.c:
10465 * gst/tcp/gstmultifdsink.c:
10466 * gst/tcp/gsttcpserversink.c:
10467 * gst/videorate/gstvideorate.c:
10468 * gst/videoscale/gstvideoscale.c:
10469 * gst/videotestsrc/gstvideotestsrc.c:
10470 * gst/volume/gstvolume.c:
10471 * sys/ximage/ximagesink.c:
10472 * sys/xvimage/xvimagesink.c:
10473 Cleanup Plugin docs. Link to signals and properties. Fix sub-section
10474 titles. Drop mentining that all our example pipelines are "simple"
10477 2008-07-07 17:25:41 +0000 Sebastian Dröge <slomo@circular-chaos.org>
10479 tests/examples/seek/Makefile.am: Fix out of tree build by adding all required CFLAGS.
10480 Original commit message from CVS:
10481 * tests/examples/seek/Makefile.am:
10482 Fix out of tree build by adding all required CFLAGS.
10484 2008-07-07 09:55:41 +0000 Sebastian Dröge <slomo@circular-chaos.org>
10486 gst/playback/gstdecodebin.c: And ref the pad before returning it again when linking to the queue failed. Otherwise we...
10487 Original commit message from CVS:
10488 * gst/playback/gstdecodebin.c: (add_raw_queue):
10489 And ref the pad before returning it again when linking to the queue
10490 failed. Otherwise we will unref the pad twice later and things break.
10492 2008-07-07 09:48:45 +0000 Sebastian Dröge <slomo@circular-chaos.org>
10494 gst/playback/gstdecodebin.c: If linking the raw pad with a queue fails, try it without a queue instead of failing com...
10495 Original commit message from CVS:
10496 * gst/playback/gstdecodebin.c: (add_raw_queue):
10497 If linking the raw pad with a queue fails, try it without a queue
10498 instead of failing completely. This should never happen.
10500 2008-07-06 23:22:12 +0000 Evgeniy Stepanov <eugeni.stepanov@gmail.com>
10502 gst/playback/gstdecodebin.c: Add a queue after a demuxer if the demuxer outputs raw data. This was done before only f...
10503 Original commit message from CVS:
10504 Patch by: Evgeniy Stepanov <eugeni dot stepanov at gmail dot com>
10505 * gst/playback/gstdecodebin.c: (add_raw_queue), (close_pad_link):
10506 Add a queue after a demuxer if the demuxer outputs raw data. This was
10507 done before only for non-raw data but is required in this case too.
10509 decodebin2 doesn't have this issue because all streams of a group
10510 go through multiqueue.
10512 2008-07-03 09:12:49 +0000 Damien Lespiau <damien.lespiau@gmail.com>
10514 gst-libs/gst/sdp/gstsdpmessage.c: Makes libgstsdp compile with mingw32 by defining the right WINVER so that getaddrin...
10515 Original commit message from CVS:
10516 Patch by: Damien Lespiau <damien dot lespiau at gmail dot com>
10517 * gst-libs/gst/sdp/gstsdpmessage.c:
10518 Makes libgstsdp compile with mingw32 by defining the right WINVER so
10519 that getaddrinfo() can be used. Fixes #541358.
10521 2008-07-01 13:22:49 +0000 Wim Taymans <wim.taymans@gmail.com>
10523 gst/videotestsrc/gstvideotestsrc.*: Cleanups, use default property values as defines.
10524 Original commit message from CVS:
10525 * gst/videotestsrc/gstvideotestsrc.c:
10526 (gst_video_test_src_class_init), (gst_video_test_src_init),
10527 (gst_video_test_src_set_property),
10528 (gst_video_test_src_get_property), (gst_video_test_src_create):
10529 * gst/videotestsrc/gstvideotestsrc.h:
10530 Cleanups, use default property values as defines.
10531 Add property to enable/disable peer buffer allocation.
10533 2008-06-30 09:46:15 +0000 Sebastian Dröge <slomo@circular-chaos.org>
10535 tests/check/: Enable unit tests on PPC again as the bugs are now fixed.
10536 Original commit message from CVS:
10537 * tests/check/elements/gdpdepay.c: (gdpdepay_suite):
10538 * tests/check/pipelines/streamheader.c: (streamheader_suite):
10539 Enable unit tests on PPC again as the bugs are now fixed.
10541 2008-06-30 09:20:59 +0000 Sebastian Dröge <slomo@circular-chaos.org>
10543 gst-libs/gst/riff/: Add support for ADPCM IMA DK3 and DK4 variant in RIFF containers.
10544 Original commit message from CVS:
10545 * gst-libs/gst/riff/riff-ids.h:
10546 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps),
10547 (gst_riff_create_audio_template_caps):
10548 Add support for ADPCM IMA DK3 and DK4 variant in RIFF containers.
10551 2008-06-30 08:29:09 +0000 Sebastian Dröge <slomo@circular-chaos.org>
10553 gst/ffmpegcolorspace/: Only set/get on the PAL8 format, ffmpegcolorspace doesn't support it on other formats. Also ad...
10554 Original commit message from CVS:
10555 * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
10556 (gst_ffmpeg_pixfmt_to_caps):
10557 * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
10558 (gst_ffmpegcsp_get_unit_size):
10559 Only set/get on the PAL8 format, ffmpegcolorspace doesn't support
10560 it on other formats. Also adjust the unit size only for that format
10561 to not include the palette. Fixes bug #540497.
10563 2008-06-29 13:45:27 +0000 Stefan Kost <ensonic@users.sourceforge.net>
10565 gst/adder/gstadder.c: Use GST_DEBUG_FUNCPTR and remove some extra vlnak lines.
10566 Original commit message from CVS:
10567 * gst/adder/gstadder.c:
10568 Use GST_DEBUG_FUNCPTR and remove some extra vlnak lines.
10570 2008-06-27 07:55:40 +0000 Stefan Kost <ensonic@users.sourceforge.net>
10572 ChangeLog: ChangeLog surgery.
10573 Original commit message from CVS:
10576 * tests/examples/seek/seek.c:
10577 Move variable into ifdef too.
10579 2008-06-27 07:42:07 +0000 Stefan Kost <ensonic@users.sourceforge.net>
10581 tests/examples/seek/seek.c: Include config.h and check if we have X. Fixes: #540334.
10582 Original commit message from CVS:
10583 * tests/examples/seek/seek.c:
10584 Include config.h and check if we have X. Fixes: #540334.
10586 2008-06-26 06:03:38 +0000 Sam Morris <sam@robots.org.to.uk>
10588 gst-libs/gst/interfaces/mixertrack.c: API: Add "index" property to GstMixerTrack to differantiate between multiple mi...
10589 Original commit message from CVS:
10590 Patch by: Sam Morris <sam at robots dot org to uk>
10591 * gst-libs/gst/interfaces/mixertrack.c:
10592 (gst_mixer_track_class_init), (gst_mixer_track_get_property),
10593 (gst_mixer_track_set_property):
10594 API: Add "index" property to GstMixerTrack to differantiate between
10595 multiple mixer tracks with the same label.
10596 * ext/alsa/gstalsamixeroptions.c: (gst_alsa_mixer_options_new):
10597 * ext/alsa/gstalsamixertrack.c: (gst_alsa_mixer_track_new):
10598 Set the "index" property of GstMixerTrack to the index given by ALSA.
10601 2008-06-25 13:15:50 +0000 Stefan Kost <ensonic@users.sourceforge.net>
10603 tests/examples/seek/: Remove libgstvideo usage. Use gtk_get_option_group instead of gtk_init().
10604 Original commit message from CVS:
10605 * tests/examples/seek/Makefile.am:
10606 * tests/examples/seek/seek.c:
10607 Remove libgstvideo usage. Use gtk_get_option_group instead of
10610 2008-06-24 16:27:35 +0000 Stefan Kost <ensonic@users.sourceforge.net>
10612 tests/check/Makefile.am: Name the test registry format neutral.
10613 Original commit message from CVS:
10614 * tests/check/Makefile.am:
10615 Name the test registry format neutral.
10617 2008-06-24 16:22:45 +0000 Stefan Kost <ensonic@users.sourceforge.net>
10619 gst/playback/gstqueue2.c: Do not double notify. Remove the unsued return value.
10620 Original commit message from CVS:
10621 * gst/playback/gstqueue2.c:
10622 Do not double notify. Remove the unsued return value.
10624 2008-06-24 16:15:26 +0000 Stefan Kost <ensonic@users.sourceforge.net>
10626 ext/alsa/gstalsamixer.c: Also consider "speaker" as a name for master volume. If that doesn't help look for the first...
10627 Original commit message from CVS:
10628 * ext/alsa/gstalsamixer.c:
10629 Also consider "speaker" as a name for master volume. If that doesn't
10630 help look for the first non-mono volume control that also has a
10633 2008-06-24 16:10:50 +0000 Stefan Kost <ensonic@users.sourceforge.net>
10635 ChangeLog: Forgot to save the ChangeLog :/
10636 Original commit message from CVS:
10638 Forgot to save the ChangeLog :/
10640 2008-06-24 16:05:06 +0000 Stefan Kost <ensonic@users.sourceforge.net>
10642 tests/examples/seek/: Embedd the xwindow.
10643 Original commit message from CVS:
10644 * tests/examples/seek/Makefile.am:
10645 * tests/examples/seek/seek.c:
10646 Embedd the xwindow.
10648 2008-06-24 01:14:40 +0000 Jan Schmidt <thaytan@mad.scientist.com>
10650 sys/ximage/ximagesink.h: When the caps change, make sure to re-draw borders in force-aspect-ratio=true mode.
10651 Original commit message from CVS:
10652 * sys/ximage/ximagesink.c (gst_ximagesink_ximage_put),
10653 (gst_ximagesink_setcaps):
10654 * sys/ximage/ximagesink.h:
10655 When the caps change, make sure to re-draw borders in
10656 force-aspect-ratio=true mode.
10657 * sys/xvimage/xvimagesink.c (gst_xvimagesink_xvimage_put):
10658 Don't clear the border_draw flag until we actually draw the border.
10659 * tests/check/Makefile.am:
10660 Ignore alsasink/src during the states test too, so it doesn't fail
10661 when running without access to the sound device.
10663 2008-06-22 18:35:27 +0000 Stefan Kost <ensonic@users.sourceforge.net>
10665 tests/examples/seek/seek.c: Fix crasher when playing a parse-launch line the 2nd time.
10666 Original commit message from CVS:
10667 * tests/examples/seek/seek.c:
10668 Fix crasher when playing a parse-launch line the 2nd time.
10670 2008-06-21 18:56:08 +0000 Thomas Vander Stichele <thomas@apestaart.org>
10672 tests/check/pipelines/oggmux.c: Properly ifdef tests to fix compilation.
10673 Original commit message from CVS:
10674 * tests/check/pipelines/oggmux.c:
10675 Properly ifdef tests to fix compilation.
10677 2008-06-21 10:25:59 +0000 Thomas Vander Stichele <thomas@apestaart.org>
10681 Original commit message from CVS:
10684 2008-06-20 18:24:24 +0000 Michael Smith <msmith@xiph.org>
10686 gst/playback/: Add get-video-pad, get-audio-pad, get-text-pad action signals to playbin2. This allows the user to get...
10687 Original commit message from CVS:
10688 * gst/playback/gstplay-marshal.list:
10689 * gst/playback/gstplaybin2.c:
10690 Add get-video-pad, get-audio-pad, get-text-pad action signals to
10691 playbin2. This allows the user to get to the selector's sinkpads, and
10692 thus inspect a range of things - caps, tags, etc.
10694 2008-06-20 17:27:03 +0000 Michael Smith <msmith@xiph.org>
10696 gst/playback/gstplaybin2.c: Use a different constant for the convert-frame signal id.
10697 Original commit message from CVS:
10698 * gst/playback/gstplaybin2.c:
10699 Use a different constant for the convert-frame signal id.
10702 2008-06-20 17:18:55 +0000 Michael Smith <msmith@xiph.org>
10704 gst/playback/: Fix a whole bunch of typos in comments and log statements.
10705 Original commit message from CVS:
10706 * gst/playback/gstplaybin2.c:
10707 * gst/playback/gstplaysink.c:
10708 Fix a whole bunch of typos in comments and log statements.
10710 2008-06-20 17:02:48 +0000 Michael Smith <msmith@xiph.org>
10712 sys/xvimage/xvimagesink.c: Don't set colour balance values on the Xv port if the user hasn't changed them (via proper...
10713 Original commit message from CVS:
10714 * sys/xvimage/xvimagesink.c:
10715 Don't set colour balance values on the Xv port if the user hasn't
10716 changed them (via properties or the interface). Avoids accumulating
10717 rounding errors for the common case.
10718 Partial fix for bug #537889.
10720 2008-06-20 16:56:18 +0000 Michael Smith <msmith@xiph.org>
10722 gst/playback/gstdecodebin2.c: Ensure decodebin2 emits 'drained' signal once, and only once, when all pads are drained.
10723 Original commit message from CVS:
10724 * gst/playback/gstdecodebin2.c:
10725 Ensure decodebin2 emits 'drained' signal once, and only once, when all
10728 2008-06-20 16:12:50 +0000 Thomas Vander Stichele <thomas@apestaart.org>
10731 apparently it's an error to specify nc -l -p 3000 - though the short usage does not make it very clear that you can d...
10732 Original commit message from CVS:
10733 apparently it's an error to specify nc -l -p 3000 - though the short usage
10734 does not make it very clear that you can drop the host arg with -l
10736 2008-06-20 09:25:44 +0000 Wim Taymans <wim.taymans@gmail.com>
10738 ext/vorbis/vorbisenc.c: Report the encoder latency. Fixes #538232.
10739 Original commit message from CVS:
10740 * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_get_latency),
10741 (gst_vorbis_enc_src_query), (gst_vorbis_enc_chain):
10742 Report the encoder latency. Fixes #538232.
10744 2008-06-20 09:19:59 +0000 Wim Taymans <wim.taymans@gmail.com>
10746 gst/playback/gstplaybin2.c: Implement the source property, emit notify when it changes in the underlying uridecodebin.
10747 Original commit message from CVS:
10748 * gst/playback/gstplaybin2.c: (gst_play_bin_get_property),
10749 (notify_source), (activate_group):
10750 Implement the source property, emit notify when it changes in the
10751 underlying uridecodebin.
10753 2008-06-20 09:14:26 +0000 Wim Taymans <wim.taymans@gmail.com>
10755 tests/examples/seek/seek.c: Free and clear the seek element list so that we don't use invalid references when seeking...
10756 Original commit message from CVS:
10757 * tests/examples/seek/seek.c: (stop_cb):
10758 Free and clear the seek element list so that we don't use invalid
10759 references when seeking after recreating a gst-launch line.
10761 2008-06-20 09:09:37 +0000 Wim Taymans <wim.taymans@gmail.com>
10763 gst-libs/gst/audio/gstbaseaudiosink.c: Report latency even if we are not live instead of hiding it.
10764 Original commit message from CVS:
10765 * gst-libs/gst/audio/gstbaseaudiosink.c:
10766 (gst_base_audio_sink_query), (gst_base_audio_sink_skew_slaving),
10767 (gst_base_audio_sink_render):
10768 Report latency even if we are not live instead of hiding it.
10769 Take ts-offset and render-delay of the basesink into account when
10770 scheduling samples.
10771 Rework the clipping code so that we can take the various offsets into
10772 account and still do correct clipping.
10774 2008-06-20 08:52:21 +0000 Jan Schmidt <thaytan@mad.scientist.com>
10776 configure.ac: Bump verion back to devel -> 0.10.20.1
10777 Original commit message from CVS:
10779 Bump verion back to devel -> 0.10.20.1
10781 2008-06-20 08:47:14 +0000 Sebastian Dröge <slomo@circular-chaos.org>
10783 gst-libs/gst/tag/tags.c: Don't increase the size of non-string image buffers by one as this might in theory confuse d...
10784 Original commit message from CVS:
10785 * gst-libs/gst/tag/tags.c: (gst_tag_image_data_to_image_buffer):
10786 Don't increase the size of non-string image buffers by one as this
10787 might in theory confuse decoders. Still increase it by one for string
10788 image buffers to append '\0'.
10790 2008-06-20 08:45:13 +0000 Antoine Tremblay <hexa00@gmail.com>
10792 gst/gdp/gstgdppay.c: Fix a buffer memleak and remove a confusing and wrong debug output.
10793 Original commit message from CVS:
10794 Patch by: Antoine Tremblay <hexa00 at gmail dot com>
10795 * gst/gdp/gstgdppay.c: (gst_gdp_pay_reset):
10796 Fix a buffer memleak and remove a confusing and wrong debug output.
10799 2008-06-19 11:25:37 +0000 Wim Taymans <wim.taymans@gmail.com>
10801 examples/app/appsink-src.c: Don't use a buffer after unreffing it.
10802 Original commit message from CVS:
10803 * examples/app/appsink-src.c: (on_new_buffer_from_source):
10804 Don't use a buffer after unreffing it.
10806 === release 0.10.20 ===
10808 2008-06-18 14:36:28 +0000 Jan Schmidt <thaytan@mad.scientist.com>
10814 * docs/plugins/gst-plugins-base-plugins.args:
10815 * docs/plugins/gst-plugins-base-plugins.hierarchy:
10816 * docs/plugins/gst-plugins-base-plugins.interfaces:
10817 * docs/plugins/gst-plugins-base-plugins.prerequisites:
10818 * docs/plugins/inspect/plugin-adder.xml:
10819 * docs/plugins/inspect/plugin-alsa.xml:
10820 * docs/plugins/inspect/plugin-audioconvert.xml:
10821 * docs/plugins/inspect/plugin-audiorate.xml:
10822 * docs/plugins/inspect/plugin-audioresample.xml:
10823 * docs/plugins/inspect/plugin-audiotestsrc.xml:
10824 * docs/plugins/inspect/plugin-cdparanoia.xml:
10825 * docs/plugins/inspect/plugin-decodebin.xml:
10826 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
10827 * docs/plugins/inspect/plugin-gdp.xml:
10828 * docs/plugins/inspect/plugin-gnomevfs.xml:
10829 * docs/plugins/inspect/plugin-libvisual.xml:
10830 * docs/plugins/inspect/plugin-ogg.xml:
10831 * docs/plugins/inspect/plugin-pango.xml:
10832 * docs/plugins/inspect/plugin-playback.xml:
10833 * docs/plugins/inspect/plugin-queue2.xml:
10834 * docs/plugins/inspect/plugin-subparse.xml:
10835 * docs/plugins/inspect/plugin-tcp.xml:
10836 * docs/plugins/inspect/plugin-theora.xml:
10837 * docs/plugins/inspect/plugin-typefindfunctions.xml:
10838 * docs/plugins/inspect/plugin-uridecodebin.xml:
10839 * docs/plugins/inspect/plugin-video4linux.xml:
10840 * docs/plugins/inspect/plugin-videorate.xml:
10841 * docs/plugins/inspect/plugin-videoscale.xml:
10842 * docs/plugins/inspect/plugin-videotestsrc.xml:
10843 * docs/plugins/inspect/plugin-volume.xml:
10844 * docs/plugins/inspect/plugin-vorbis.xml:
10845 * docs/plugins/inspect/plugin-ximagesink.xml:
10846 * docs/plugins/inspect/plugin-xvimagesink.xml:
10847 * gst-plugins-base.doap:
10849 * win32/common/config.h:
10851 Original commit message from CVS:
10854 2008-06-18 14:32:12 +0000 Jan Schmidt <thaytan@mad.scientist.com>
10883 Original commit message from CVS:
10886 2008-06-18 06:31:11 +0000 Stefan Kost <ensonic@users.sourceforge.net>
10888 Fix gtk-doc warnings. Also don't misuse api-doc comments for normal comments.
10889 Original commit message from CVS:
10890 * docs/plugins/gst-plugins-bad-plugins-sections.txt:
10891 * examples/app/appsrc-ra.c:
10892 * examples/app/appsrc-seekable.c:
10893 * examples/app/appsrc-stream.c:
10894 * examples/app/appsrc-stream2.c:
10895 * ext/directfb/dfbvideosink.h:
10896 * ext/metadata/gstbasemetadata.c:
10897 * ext/metadata/gstbasemetadata.h:
10898 * ext/metadata/metadata.c:
10899 * ext/metadata/metadataexif.c:
10900 * ext/theora/theoradec.h:
10901 * gst/deinterlace2/gstdeinterlace2.h:
10902 * gst/deinterlace2/tvtime/speedy.c:
10903 * gst/deinterlace2/tvtime/speedy.h:
10904 * gst/deinterlace2/tvtime/vfir.c:
10905 Fix gtk-doc warnings. Also don't misuse api-doc comments for normal
10908 2008-06-16 14:11:36 +0000 Andy Wingo <wingo@pobox.com>
10910 * gst-libs/gst/app/gstappsrc.c:
10911 gst-libs/gst/app/gstappsrc.c (gst_app_src_set_max_bytes)
10912 Original commit message from CVS:
10913 2008-06-16 Andy Wingo <wingo@pobox.com>
10914 * gst-libs/gst/app/gstappsrc.c (gst_app_src_set_max_bytes)
10915 (gst_app_src_get_max_bytes, gst_app_src_push_buffer): Use
10916 G_GUINT64_FORMAT. Avoid overflow in get_max_bytes().
10918 2008-06-16 07:30:32 +0000 Stefan Kost <ensonic@users.sourceforge.net>
10920 Final round of doc updates.
10921 Original commit message from CVS:
10922 * gst/rtpmanager/gstrtpjitterbuffer.c:
10923 * gst/speed/gstspeed.c:
10924 * gst/speexresample/gstspeexresample.c:
10925 * gst/videosignal/gstvideoanalyse.c:
10926 * gst/videosignal/gstvideodetect.c:
10927 * gst/videosignal/gstvideomark.c:
10928 * sys/dvb/gstdvbsrc.c:
10929 * sys/oss4/oss4-mixer.c:
10930 * sys/oss4/oss4-sink.c:
10931 * sys/oss4/oss4-source.c:
10932 * sys/wininet/gstwininetsrc.c:
10933 Final round of doc updates.
10935 2008-06-13 11:59:21 +0000 Stefan Kost <ensonic@users.sourceforge.net>
10937 docs/plugins/: docs/plugins/inspect/plugin-mythtv.xml
10938 Original commit message from CVS:
10939 * docs/plugins/Makefile.am:
10940 * docs/plugins/gst-plugins-bad-plugins-docs.sgml:
10941 * docs/plugins/gst-plugins-bad-plugins-sections.txt:
10942 * docs/plugins/gst-plugins-bad-plugins.args:
10943 * docs/plugins/gst-plugins-bad-plugins.hierarchy:
10944 * docs/plugins/gst-plugins-bad-plugins.interfaces:
10945 * docs/plugins/gst-plugins-bad-plugins.prerequisites:
10946 * docs/plugins/gst-plugins-bad-plugins.signals:
10947 * docs/plugins/inspect/plugin-alsaspdif.xml:
10948 * docs/plugins/inspect/plugin-amrwb.xml:
10949 * docs/plugins/inspect/plugin-app.xml:
10950 * docs/plugins/inspect/plugin-bayer.xml:
10951 * docs/plugins/inspect/plugin-bz2.xml:
10952 * docs/plugins/inspect/plugin-cdaudio.xml:
10953 * docs/plugins/inspect/plugin-cdxaparse.xml:
10954 * docs/plugins/inspect/plugin-dtsdec.xml:
10955 * docs/plugins/inspect/plugin-dvb.xml:
10956 * docs/plugins/inspect/plugin-dvdspu.xml:
10957 * docs/plugins/inspect/plugin-faac.xml:
10958 * docs/plugins/inspect/plugin-faad.xml:
10959 * docs/plugins/inspect/plugin-fbdevsink.xml:
10960 * docs/plugins/inspect/plugin-festival.xml:
10961 * docs/plugins/inspect/plugin-filter.xml:
10962 * docs/plugins/inspect/plugin-flvdemux.xml:
10963 * docs/plugins/inspect/plugin-freeze.xml:
10964 * docs/plugins/inspect/plugin-gsm.xml:
10965 * docs/plugins/inspect/plugin-gstinterlace.xml:
10966 * docs/plugins/inspect/plugin-gstrtpmanager.xml:
10967 * docs/plugins/inspect/plugin-h264parse.xml:
10968 * docs/plugins/inspect/plugin-interleave.xml:
10969 * docs/plugins/inspect/plugin-jack.xml:
10970 * docs/plugins/inspect/plugin-ladspa.xml:
10971 * docs/plugins/inspect/plugin-metadata.xml:
10972 * docs/plugins/inspect/plugin-mms.xml:
10973 * docs/plugins/inspect/plugin-modplug.xml:
10974 * docs/plugins/inspect/plugin-mpeg2enc.xml:
10975 * docs/plugins/inspect/plugin-mpeg4videoparse.xml:
10976 * docs/plugins/inspect/plugin-mpegtsparse.xml:
10977 * docs/plugins/inspect/plugin-mpegvideoparse.xml:
10978 * docs/plugins/inspect/plugin-musepack.xml:
10979 * docs/plugins/inspect/plugin-musicbrainz.xml:
10980 * docs/plugins/inspect/plugin-mve.xml:
10981 * docs/plugins/inspect/plugin-mythtv.xml
10982 * docs/plugins/inspect/plugin-nas.xml:
10983 * docs/plugins/inspect/plugin-neon.xml:
10984 * docs/plugins/inspect/plugin-nsfdec.xml:
10985 * docs/plugins/inspect/plugin-nuvdemux.xml:
10986 * docs/plugins/inspect/plugin-oss4.xml
10987 * docs/plugins/inspect/plugin-rawparse.xml:
10988 * docs/plugins/inspect/plugin-real.xml:
10989 * docs/plugins/inspect/plugin-replaygain.xml:
10990 * docs/plugins/inspect/plugin-rfbsrc.xml:
10991 * docs/plugins/inspect/plugin-sdl.xml:
10992 * docs/plugins/inspect/plugin-sdp.xml:
10993 * docs/plugins/inspect/plugin-selector.xml:
10994 * docs/plugins/inspect/plugin-sndfile.xml:
10995 * docs/plugins/inspect/plugin-soundtouch.xml:
10996 * docs/plugins/inspect/plugin-spcdec.xml:
10997 * docs/plugins/inspect/plugin-speed.xml:
10998 * docs/plugins/inspect/plugin-speexresample.xml:
10999 * docs/plugins/inspect/plugin-stereo.xml:
11000 * docs/plugins/inspect/plugin-subenc.xml
11001 * docs/plugins/inspect/plugin-timidity.xml:
11002 * docs/plugins/inspect/plugin-tta.xml:
11003 * docs/plugins/inspect/plugin-vcdsrc.xml:
11004 * docs/plugins/inspect/plugin-videosignal.xml:
11005 * docs/plugins/inspect/plugin-vmnc.xml:
11006 * docs/plugins/inspect/plugin-wildmidi.xml:
11007 * docs/plugins/inspect/plugin-x264.xml:
11008 * docs/plugins/inspect/plugin-xvid.xml:
11009 * docs/plugins/inspect/plugin-y4menc.xml:
11010 * ext/amrwb/gstamrwbdec.c:
11011 * ext/amrwb/gstamrwbenc.c:
11012 * ext/amrwb/gstamrwbparse.c:
11013 * ext/dc1394/gstdc1394.c:
11014 * ext/directfb/dfbvideosink.c:
11015 * ext/ivorbis/vorbisdec.c:
11016 * ext/jack/gstjackaudiosink.c:
11017 * ext/mpeg2enc/gstmpeg2enc.cc:
11018 * ext/mplex/gstmplex.cc:
11019 * ext/musicbrainz/gsttrm.c:
11020 * ext/mythtv/gstmythtvsrc.c:
11021 * ext/theora/theoradec.c:
11022 * ext/timidity/gsttimidity.c:
11023 * ext/timidity/gstwildmidi.c:
11024 * gst-libs/gst/app/gstappsink.c:
11025 * gst/deinterlace/gstdeinterlace.c:
11026 * gst/dvdspu/gstdvdspu.c:
11027 * gst/festival/gstfestival.c:
11028 * gst/freeze/gstfreeze.c:
11029 * gst/interleave/deinterleave.c:
11030 * gst/interleave/interleave.c:
11031 * gst/modplug/gstmodplug.cc:
11032 * gst/nuvdemux/gstnuvdemux.c:
11033 Add missing elements to docs. Fix doc-markup: use convinience syntax
11034 for examples (produces valid docbook), add several refsec2 when we
11035 have several titles. Fix some types.
11037 2008-06-12 15:47:03 +0000 Wim Taymans <wim.taymans@gmail.com>
11039 examples/app/: Add beefed up example app from bug #413418. It now also uses appsink instead of fakesink for more ulti...
11040 Original commit message from CVS:
11041 * examples/app/.cvsignore:
11042 * examples/app/Makefile.am:
11043 * examples/app/appsink-src.c: (on_new_buffer_from_source),
11044 (on_source_message), (on_sink_message), (main):
11045 Add beefed up example app from bug #413418. It now also uses appsink
11046 instead of fakesink for more ultimate coolness.
11047 * gst-libs/gst/app/gstappsrc.c: (gst_app_src_class_init),
11048 (gst_app_src_init), (gst_app_src_set_property),
11049 (gst_app_src_get_property), (gst_app_src_unlock),
11050 (gst_app_src_unlock_stop), (gst_app_src_create),
11051 (gst_app_src_set_max_bytes), (gst_app_src_push_buffer),
11052 (gst_app_src_end_of_stream):
11053 * gst-libs/gst/app/gstappsrc.h:
11054 Add block property to allow push based implementation to block when we
11055 fill up the appsrc queues.
11056 Emit the enough-data signal while releasing our lock.
11058 2008-06-12 14:50:27 +0000 Stefan Kost <ensonic@users.sourceforge.net>
11060 examples/app/.cvsignore: Ignore more.
11061 Original commit message from CVS:
11062 * examples/app/.cvsignore:
11065 2008-06-12 14:49:15 +0000 Stefan Kost <ensonic@users.sourceforge.net>
11067 Do not use short_description in section docs for elements. We extract them from element details and there will be war...
11068 Original commit message from CVS:
11069 * ext/dc1394/gstdc1394.c:
11070 * ext/ivorbis/vorbisdec.c:
11071 * ext/jack/gstjackaudiosink.c:
11072 * ext/metadata/gstmetadatademux.c:
11073 * ext/mythtv/gstmythtvsrc.c:
11074 * ext/theora/theoradec.c:
11075 * gst-libs/gst/app/gstappsink.c:
11076 * gst/bayer/gstbayer2rgb.c:
11077 * gst/deinterlace/gstdeinterlace.c:
11078 * gst/rawparse/gstaudioparse.c:
11079 * gst/rawparse/gstvideoparse.c:
11080 * gst/rtpmanager/gstrtpbin.c:
11081 * gst/rtpmanager/gstrtpclient.c:
11082 * gst/rtpmanager/gstrtpjitterbuffer.c:
11083 * gst/rtpmanager/gstrtpptdemux.c:
11084 * gst/rtpmanager/gstrtpsession.c:
11085 * gst/rtpmanager/gstrtpssrcdemux.c:
11086 * gst/selector/gstinputselector.c:
11087 * gst/selector/gstoutputselector.c:
11088 * gst/videosignal/gstvideoanalyse.c:
11089 * gst/videosignal/gstvideodetect.c:
11090 * gst/videosignal/gstvideomark.c:
11091 * sys/oss4/oss4-mixer.c:
11092 * sys/oss4/oss4-sink.c:
11093 * sys/oss4/oss4-source.c:
11094 Do not use short_description in section docs for elements. We extract
11095 them from element details and there will be warnings if they differ.
11096 Also fixing up the ChangeLog order.
11098 2008-06-11 21:17:01 +0000 Jan Schmidt <thaytan@mad.scientist.com>
11100 configure.ac: 0.10.19.3 pre-release
11101 Original commit message from CVS:
11103 0.10.19.3 pre-release
11105 2008-06-11 20:13:00 +0000 David Schleef <ds@schleef.org>
11107 gst-libs/gst/rtsp/gstrtspconnection.c: Fix build on win32.
11108 Original commit message from CVS:
11109 * gst-libs/gst/rtsp/gstrtspconnection.c:
11110 Fix build on win32.
11111 Patch By: David Schleef <ds@schleef.org>
11114 2008-06-11 09:35:51 +0000 Sebastian Dröge <slomo@circular-chaos.org>
11116 ext/gio/gstgiobasesrc.*: Try to read the requested number of bytes, even if the first read returns less than requeste...
11117 Original commit message from CVS:
11118 * ext/gio/gstgiobasesrc.c: (gst_gio_base_src_finalize),
11119 (gst_gio_base_src_create):
11120 * ext/gio/gstgiobasesrc.h:
11121 Try to read the requested number of bytes, even if the first
11122 read returns less than requested, until nothing is read anymore
11123 or we have the requested amount of bytes. This fixes playback of
11124 files via Samba as Samba only allows to read 64k at once.
11125 Implement a caching algorithm that makes sure that we read at
11126 least 4k of data every time. Some elements will try to read a few
11127 bytes, then seek, read again a few bytes and so on and this is
11128 painfully slow as every operation has to go over DBus if GVfs is
11130 Fixes bug #536849 and #536848.
11131 * ext/gio/gstgiosrc.c: (gst_gio_src_class_init),
11132 (gst_gio_src_check_get_range):
11133 Override check_get_range() to blacklist http/https URIs
11134 and whitelist file URIs. More to be added on demand.
11136 2008-06-06 16:50:51 +0000 Wim Taymans <wim.taymans@gmail.com>
11138 examples/app/: Added 3 more example application for using appsrc in random-access mode, pull-mode streaming and pull ...
11139 Original commit message from CVS:
11140 * examples/app/Makefile.am:
11141 * examples/app/appsrc-ra.c: (feed_data), (seek_data),
11142 (found_source), (bus_message), (main):
11143 * examples/app/appsrc-seekable.c: (feed_data), (seek_data),
11144 (found_source), (bus_message), (main):
11145 * examples/app/appsrc-stream2.c: (feed_data), (found_source),
11146 (bus_message), (main):
11147 Added 3 more example application for using appsrc in random-access mode,
11148 pull-mode streaming and pull mode seekable.
11149 * gst-libs/gst/app/gstappsrc.c: (gst_app_src_class_init),
11150 (gst_app_src_start), (gst_app_src_do_get_size),
11151 (gst_app_src_create):
11152 * gst-libs/gst/app/gstappsrc.h:
11153 Make stream-type property writable.
11154 Unset flushing when starting so that we reuse appsrc.
11155 Inform basesrc about the configured size.
11156 Emit seek-data signal when we are going to a different offset in
11157 random-access mode.
11159 2008-06-06 14:19:54 +0000 Wim Taymans <wim.taymans@gmail.com>
11161 examples/app/appsrc-stream.c: Use deep-notify until we can depend on a playbin2 with support for the source property.
11162 Original commit message from CVS:
11163 * examples/app/appsrc-stream.c: (found_source), (main):
11164 Use deep-notify until we can depend on a playbin2 with support for the
11167 2008-06-05 16:38:50 +0000 Wim Taymans <wim.taymans@gmail.com>
11169 examples/app/: Added an example on how to use appsrc in playbin in streaming mode from an mmapped file.
11170 Original commit message from CVS:
11171 * examples/app/.cvsignore:
11172 * examples/app/Makefile.am:
11173 * examples/app/appsrc-stream.c: (read_data), (start_feed),
11174 (stop_feed), (found_source), (bus_message), (main):
11175 Added an example on how to use appsrc in playbin in streaming mode from
11177 * examples/app/appsrc_ex.c: (main):
11178 Set pipeline to NULL to free queued buffers.
11179 * gst-libs/gst/app/gstapp-marshal.list:
11180 * gst-libs/gst/app/gstappsrc.c: (stream_type_get_type), (_do_init),
11181 (gst_app_src_class_init), (gst_app_src_init),
11182 (gst_app_src_flush_queued), (gst_app_src_dispose),
11183 (gst_app_src_set_property), (gst_app_src_get_property),
11184 (gst_app_src_unlock), (gst_app_src_unlock_stop),
11185 (gst_app_src_start), (gst_app_src_stop), (gst_app_src_is_seekable),
11186 (gst_app_src_check_get_range), (gst_app_src_do_seek),
11187 (gst_app_src_create), (gst_app_src_set_stream_type),
11188 (gst_app_src_get_stream_type), (gst_app_src_set_max_bytes),
11189 (gst_app_src_get_max_bytes), (gst_app_src_push_buffer),
11190 (gst_app_src_end_of_stream), (gst_app_src_uri_get_type),
11191 (gst_app_src_uri_get_protocols), (gst_app_src_uri_get_uri),
11192 (gst_app_src_uri_set_uri), (gst_app_src_uri_handler_init):
11193 * gst-libs/gst/app/gstappsrc.h:
11194 Measure max queue size in bytes instead.
11195 Add support for 3 modes of operation, streaming, seekable and
11196 random-access, making basesrc handle the scheduling modes for each.
11197 Add appsrc:// uri handler so that automatic plugging can be done from
11198 playbin2 or uridecodebin, for example.
11199 Added support for custom segment formats.
11200 Add support for push and pull based operations from the application.
11201 Expand the methods so that errors can be detected.
11202 Flush the queued buffers on seeks and when shutting down.
11203 Add signals to inform the app that a seek must happen.
11205 2008-06-05 09:47:23 +0000 Jan Schmidt <thaytan@mad.scientist.com>
11207 configure.ac: 0.10.19.2 pre-release
11208 Original commit message from CVS:
11210 0.10.19.2 pre-release
11212 2008-06-04 21:48:27 +0000 Jan Schmidt <thaytan@mad.scientist.com>
11214 win32/common/: Add new API functions to the dll exports
11215 Original commit message from CVS:
11216 * win32/common/libgstrtsp.def:
11217 * win32/common/libgsttag.def:
11218 Add new API functions to the dll exports
11220 2008-06-04 17:42:38 +0000 Michael Smith <msmith@xiph.org>
11222 gst/playback/gstplaybasebin.c: Disconnect signals from decodebins we created before we remove it from playbin, to avo...
11223 Original commit message from CVS:
11224 * gst/playback/gstplaybasebin.c:
11225 Disconnect signals from decodebins we created before we remove it from
11226 playbin, to avoid crashes if the decodebin is eventually disposed after
11227 the playbin itself (possible if the app takes a reference on the
11231 2008-06-04 17:12:40 +0000 Tim-Philipp Müller <tim@centricular.net>
11233 gst/typefind/gsttypefindfunctions.c: Bunch of small clean-ups: use gst_type_find_suggest_simple(); don't copy caps fo...
11234 Original commit message from CVS:
11235 * gst/typefind/gsttypefindfunctions.c: (aac_type_find),
11236 (mp3_type_find), (musepack_type_find), (MULTIPART_MAX_HEADER_SIZE),
11237 (mpeg_sys_type_find), (mpeg_ts_type_find), (mpeg4_video_type_find),
11238 (h264_video_type_find), (mpeg_video_stream_type_find),
11239 (dv_type_find), (mmsh_type_find):
11240 Bunch of small clean-ups: use gst_type_find_suggest_simple(); don't
11241 copy caps for no good reason (this may be desirable to make it easier
11242 to detect leaks, but then it should probably be done for all caps
11243 in the typefinder somewhere).
11245 2008-06-04 16:06:49 +0000 Peter Kjellerstedt <pkj@axis.com>
11247 tests/check/Makefile.am: Do not try to run the check tests for subparse unless it has been built.
11248 Original commit message from CVS:
11249 * tests/check/Makefile.am:
11250 Do not try to run the check tests for subparse unless it has been
11253 2008-06-04 16:00:26 +0000 Peter Kjellerstedt <pkj@axis.com>
11255 tests/check/pipelines/streamheader.c: Do not try to run a test which requires vorbisenc unless we have actually built...
11256 Original commit message from CVS:
11257 * tests/check/pipelines/streamheader.c: (buffer_probe_cb),
11258 (test_multifdsink_gdp_vorbisenc), (streamheader_suite):
11259 Do not try to run a test which requires vorbisenc unless we have
11262 2008-06-04 11:53:53 +0000 Peter Kjellerstedt <pkj@axis.com>
11264 gst-libs/gst/rtsp/gstrtspconnection.*: Add a couple of missing argument guards.
11265 Original commit message from CVS:
11266 * gst-libs/gst/rtsp/gstrtspconnection.c:
11267 (gst_rtsp_connection_set_auth), (gst_rtsp_connection_set_auth_param),
11268 (gst_rtsp_connection_clear_auth_params),
11269 (gst_rtsp_connection_set_qos_dscp), (gst_rtsp_connection_get_ip):
11270 * gst-libs/gst/rtsp/gstrtspconnection.h:
11271 Add a couple of missing argument guards.
11272 Add a way of setting the DSCP for an RTSP connection.
11273 Add an accessor method for the ip member of GstRTSPConnection as all
11274 members are supposed to be private.
11276 2008-06-04 11:33:23 +0000 Peter Kjellerstedt <pkj@axis.com>
11278 gst/tcp/gstmultifdsink.c: Fixed accidental use of IPv4 options for all IPv6 addresses.
11279 Original commit message from CVS:
11280 * gst/tcp/gstmultifdsink.c: (setup_dscp_client):
11281 Fixed accidental use of IPv4 options for all IPv6 addresses.
11283 2008-06-04 10:18:42 +0000 Tim-Philipp Müller <tim@centricular.net>
11285 gst-libs/gst/interfaces/mixertrack.h: Document mixer track flags.
11286 Original commit message from CVS:
11287 * gst-libs/gst/interfaces/mixertrack.h:
11288 Document mixer track flags.
11290 2008-06-04 05:58:38 +0000 Antoine Tremblay <hexa00@gmail.com>
11292 gst/gdp/gstgdppay.c: Don't set caps on the buffers that contain a copy of the buffer including the caps of them resul...
11293 Original commit message from CVS:
11294 Patch by: Antoine Tremblay <hexa00 at gmail dot com>
11295 * gst/gdp/gstgdppay.c: (gst_gdp_pay_reset_streamheader):
11296 Don't set caps on the buffers that contain a copy of the buffer
11297 including the caps of them resulting in an always increasing refcount
11298 of the caps and insanely large caps. Instead include a buffer without
11299 caps in the new caps. Fixes bug #536475.
11301 2008-06-04 05:44:06 +0000 Sebastian Dröge <slomo@circular-chaos.org>
11303 gst/videoscale/gstvideoscale.c: Transform a given PAR to a range on the struct with the generic height/width instead ...
11304 Original commit message from CVS:
11305 * gst/videoscale/gstvideoscale.c: (gst_video_scale_transform_caps):
11306 Transform a given PAR to a range on the struct with the generic
11307 height/width instead of the struct with the possibly restricted
11310 2008-06-04 04:24:27 +0000 Sebastian Dröge <slomo@circular-chaos.org>
11312 gst/videoscale/gstvideoscale.c: Prefer the given format if it contains something stricter than [1,MAX] for height or ...
11313 Original commit message from CVS:
11314 * gst/videoscale/gstvideoscale.c: (gst_video_scale_transform_caps):
11315 Prefer the given format if it contains something stricter than [1,MAX]
11316 for height or width and only put a structure that requires rescaling
11317 as second. This makes it possible to use videoscale in pipelines where
11318 the source can actually produce the wanted height/width but usually
11319 selects a different one from the requested.
11321 2008-06-03 20:01:58 +0000 John Millikin <jmillikin@gmail.com>
11323 gst-libs/gst/tag/gstvorbistag.c: Retrieve COVERART tags from vorbis comments (#512333)
11324 Original commit message from CVS:
11325 Based on patch by: John Millikin <jmillikin gmail com>
11326 * gst-libs/gst/tag/gstvorbistag.c: (tag_matches), (gst_vorbis_tag_add),
11327 (gst_vorbis_tag_add_coverart):
11328 Retrieve COVERART tags from vorbis comments (#512333)
11330 2008-06-03 19:44:48 +0000 Tim-Philipp Müller <tim@centricular.net>
11332 gst-libs/gst/tag/: Don't forget to add new enum value here too (should probably use glib-mkenums here...).
11333 Original commit message from CVS:
11334 * gst-libs/gst/tag/tag.h:
11335 * gst-libs/gst/tag/tags.c: (register_tag_image_type_enum):
11336 Don't forget to add new enum value here too (should probably use
11337 glib-mkenums here...).
11339 2008-06-03 19:29:06 +0000 Tim-Philipp Müller <tim@centricular.net>
11341 gst-libs/gst/tag/: API: add gst_tag_image_data_to_image_buffer()
11342 Original commit message from CVS:
11343 * gst-libs/gst/tag/gstid3tag.c: (gst_tag_list_add_id3_image):
11344 * gst-libs/gst/tag/tag.h: (GST_TAG_IMAGE_TYPE_NONE),
11345 * gst-libs/gst/tag/tags.c: (register_tag_image_type_enum),
11346 (gst_tag_image_type_get_type), (gst_tag_image_type_is_valid),
11347 (gst_tag_image_data_to_image_buffer):
11348 Add two utility functions to avoid code duplication (#512333):
11349 API: add gst_tag_image_data_to_image_buffer()
11350 API: add gst_tag_list_add_id3_image()
11352 2008-06-03 08:54:29 +0000 Sebastian Dröge <slomo@circular-chaos.org>
11354 win32/common/libgstaudio.def: Add gst_audio_check_channel_positions() to the exported symbols.
11355 Original commit message from CVS:
11356 * win32/common/libgstaudio.def:
11357 Add gst_audio_check_channel_positions() to the exported symbols.
11359 2008-06-03 08:48:32 +0000 Sebastian Dröge <slomo@circular-chaos.org>
11361 API: Make gst_audio_check_channel_positions() public.
11362 Original commit message from CVS:
11363 * docs/libs/gst-plugins-base-libs-sections.txt:
11364 * gst-libs/gst/audio/multichannel.c:
11365 (gst_audio_check_channel_positions):
11366 * gst-libs/gst/audio/multichannel.h:
11367 API: Make gst_audio_check_channel_positions() public.
11368 * tests/check/libs/audio.c: (GST_START_TEST):
11369 Add some simple checks for gst_audio_check_channel_positions().
11371 2008-06-02 20:09:14 +0000 Tim-Philipp Müller <tim@centricular.net>
11373 sys/v4l/v4l_calls.c: minrange and maxrange are scaled according to the frequency multiplier.
11374 Original commit message from CVS:
11375 * sys/v4l/v4l_calls.c: (gst_v4l_get_chan_names):
11376 minrange and maxrange are scaled according to the frequency
11379 2008-06-02 18:37:02 +0000 Tim-Philipp Müller <tim@centricular.net>
11381 ext/pango/: Use gstvideo functions to calculate strides and plane offsets. Fixes rendering issue ('ghost' images of t...
11382 Original commit message from CVS:
11383 * ext/pango/Makefile.am:
11384 * ext/pango/gsttextoverlay.c: (gst_text_overlay_shade_y),
11385 (gst_text_overlay_blit_yuv420), (gst_text_overlay_push_frame):
11386 Use gstvideo functions to calculate strides and plane offsets. Fixes
11387 rendering issue ('ghost' images of the text on the chroma planes)
11388 with widths or heights that are not multiples of 8 (#506659 and
11389 probably also #485729).
11390 * tests/icles/test-textoverlay.c: (show_text), (test_textoverlay),
11392 Test with odd height/width too.
11394 2008-06-02 12:20:35 +0000 Sebastian Dröge <slomo@circular-chaos.org>
11396 gst/adder/gstadder.c: When using gst_element_iterate_pads() one has to unref every pad after usage.
11397 Original commit message from CVS:
11398 * gst/adder/gstadder.c: (gst_adder_query_duration),
11399 (gst_adder_query_latency):
11400 When using gst_element_iterate_pads() one has to unref every pad
11403 2008-05-31 19:57:57 +0000 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
11405 gst-libs/gst/audio/gstbaseaudiosrc.c: Add a gtk-doc chunk for the new properties to have a Since: indication.
11406 Original commit message from CVS:
11407 * gst-libs/gst/audio/gstbaseaudiosrc.c:
11408 (gst_base_audio_src_class_init):
11409 Add a gtk-doc chunk for the new properties to have a Since: indication.
11411 2008-05-31 19:50:59 +0000 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
11414 ChangeLog surgery, mark API change
11415 Original commit message from CVS:
11416 ChangeLog surgery, mark API change
11418 2008-05-31 18:10:47 +0000 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
11420 gst-libs/gst/audio/gstbaseaudiosrc.c: Provide readable actual-buffer-time and actual-latency-time properties that ref...
11421 Original commit message from CVS:
11422 * gst-libs/gst/audio/gstbaseaudiosrc.c:
11423 (gst_base_audio_src_class_init), (gst_base_audio_src_dispose),
11424 (gst_base_audio_src_get_property), (gst_base_audio_src_setcaps),
11425 (gst_base_audio_src_change_state):
11426 Provide readable actual-buffer-time and actual-latency-time properties
11427 that reflect the configured ringbuffer values. Fixes #524724.
11429 2008-05-30 15:29:20 +0000 Wim Taymans <wim.taymans@gmail.com>
11431 gst-libs/gst/rtp/gstbasertppayload.c: Simply converting the running time into an RTP timestamp by scaling it based on...
11432 Original commit message from CVS:
11433 * gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_push),
11434 (gst_basertppayload_change_state):
11435 Simply converting the running time into an RTP timestamp by scaling it
11436 based on the clock-rate is good enough for making an RTP timestamp. This
11437 has the added benefit that we can later on expose a property with the
11438 RTP timestamp of running time 0, as is needed for RTSP servers to
11439 generate the response of the PLAY request.
11441 2008-05-30 08:42:17 +0000 Sebastian Dröge <slomo@circular-chaos.org>
11443 gst/audioconvert/gstaudioconvert.c: Allow up to 11 positioned channels now that audioconvert can handle this but add ...
11444 Original commit message from CVS:
11445 * gst/audioconvert/gstaudioconvert.c:
11446 (structure_has_fixed_channel_positions),
11447 (gst_audio_convert_transform_caps):
11448 Allow up to 11 positioned channels now that audioconvert can handle
11449 this but add no default positions for > 8 channels.
11450 * tests/check/elements/audioconvert.c: (GST_START_TEST):
11451 Add some unit tests for the above change: Test conversion of
11452 11 positioned channels to stereo and the other way around, test
11453 conversion of 15 unpositioned channels in different ways.
11455 2008-05-29 19:45:40 +0000 Sebastian Dröge <slomo@circular-chaos.org>
11457 win32/common/libgstaudio.def: Add gst_audio_clock_reset to the list of exported symbols.
11458 Original commit message from CVS:
11459 * win32/common/libgstaudio.def:
11460 Add gst_audio_clock_reset to the list of exported symbols.
11462 2008-05-29 19:37:47 +0000 Sebastian Dröge <slomo@circular-chaos.org>
11464 tests/check/elements/vorbisdec.c: Remove wrong_channels_identification_header unit test as we now support 7 (and more...
11465 Original commit message from CVS:
11466 * tests/check/elements/vorbisdec.c: (vorbisdec_suite):
11467 Remove wrong_channels_identification_header unit test as we now
11468 support 7 (and more channels).
11470 2008-05-29 12:17:16 +0000 Sebastian Dröge <slomo@circular-chaos.org>
11472 gst/audioconvert/gstchannelmix.c: If mixing left or right to center (or the other way around) only take the complete ...
11473 Original commit message from CVS:
11474 * gst/audioconvert/gstchannelmix.c:
11475 (gst_channel_mix_fill_one_other):
11476 If mixing left or right to center (or the other way around) only take
11477 the complete value if we don't already have the original position in
11480 2008-05-29 11:34:09 +0000 Sebastian Dröge <slomo@circular-chaos.org>
11482 gst-libs/gst/audio/multichannel.c: Allow rear center together with rear left/right and other previously conflicting c...
11483 Original commit message from CVS:
11484 * gst-libs/gst/audio/multichannel.c:
11485 (gst_audio_check_channel_positions),
11486 (gst_audio_set_structure_channel_positions_list),
11487 (gst_audio_fixate_channel_positions):
11488 Allow rear center together with rear left/right and other previously
11489 conflicting channel positions. The reason why they weren't allowed
11490 was the channel mixing implementation in audioconvert.
11491 Also take this into account when fixing channel layouts.
11492 Allow setting channel positions for 1/2 channels when using
11493 gst_audio_set_structure_channel_position().
11494 * gst/audioconvert/gstchannelmix.c:
11495 (gst_channel_mix_fill_compatible), (gst_channel_mix_detect_pos),
11496 (gst_channel_mix_fill_one_other), (gst_channel_mix_fill_others),
11497 (gst_channel_mix_fill_special), (gst_channel_mix_fill_matrix):
11498 Major rewrite of the channel mixing.
11499 We now allow previously conflicting channel positions to appear
11500 together (rear center and rear left/right for example).
11502 Rework the way channels are mixed together to take more possible
11503 channel positions into account, properly mix from/to side channels
11504 and don't assume that either center, left&right or nothing of a
11505 specific position is available anymore.
11506 * tests/check/elements/audioconvert.c: (GST_START_TEST):
11507 Adjust unit tests with non-standard 1/2 channel layouts to the more
11508 correct new behaviour.
11509 Add a unit test for 5.1->Stereo downmixing.
11511 2008-05-29 07:02:50 +0000 Sebastian Dröge <slomo@circular-chaos.org>
11513 ext/vorbis/: Add sane defaults for the 7 and 8 channel layouts as those are undefined in the Vorbis spec. Use NONE ch...
11514 Original commit message from CVS:
11515 * ext/vorbis/vorbisdec.c: (vorbis_handle_identification_packet):
11516 * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_generate_sink_caps):
11517 Add sane defaults for the 7 and 8 channel layouts as those are
11518 undefined in the Vorbis spec. Use NONE channel layouts when decoding
11519 more than 8 channels instead of erroring out. Fixes bug #535356.
11521 2008-05-28 16:10:20 +0000 Wim Taymans <wim.taymans@gmail.com>
11523 Add theoraparse to the docs and fix some docs.
11524 Original commit message from CVS:
11525 * docs/plugins/Makefile.am:
11526 * docs/plugins/gst-plugins-base-plugins-docs.sgml:
11527 * docs/plugins/gst-plugins-base-plugins-sections.txt:
11528 * ext/theora/theoraparse.c:
11529 Add theoraparse to the docs and fix some docs.
11531 2008-05-28 15:48:33 +0000 Wim Taymans <wim.taymans@gmail.com>
11533 gst-libs/gst/cdda/gstcddabasesrc.c: Fix EOS condition and track addition check, the track.end sector is included in t...
11534 Original commit message from CVS:
11535 * gst-libs/gst/cdda/gstcddabasesrc.c:
11536 (gst_cdda_base_src_add_track), (gst_cdda_base_src_create):
11537 Fix EOS condition and track addition check, the track.end sector is
11538 included in the track. Fixes #533265.
11540 2008-05-28 14:49:24 +0000 Mark Nauwelaerts <manauw@skynet.be>
11542 gst/videorate/gstvideorate.*: React (more) to NEWSEGMENT
11543 Original commit message from CVS:
11544 Patch by: Mark Nauwelaerts <manauw at skynet be>
11545 * gst/videorate/gstvideorate.c: (gst_video_rate_reset),
11546 (gst_video_rate_flush_prev), (gst_video_rate_event),
11547 (gst_video_rate_chain):
11548 * gst/videorate/gstvideorate.h:
11549 React (more) to NEWSEGMENT
11550 Small adjustment in timestamp calculation to prevent mismatches
11553 2008-05-28 11:31:44 +0000 Tim-Philipp Müller <tim@centricular.net>
11555 tests/examples/seek/seek.c: Initialise error to NULL as we should.
11556 Original commit message from CVS:
11557 * tests/examples/seek/seek.c: (make_parselaunch_pipeline):
11558 Initialise error to NULL as we should.
11560 2008-05-28 08:14:47 +0000 Sebastian Dröge <slomo@circular-chaos.org>
11562 gst/adder/gstadder.c: Implement latency query.
11563 Original commit message from CVS:
11564 * gst/adder/gstadder.c: (gst_adder_query_duration),
11565 (gst_adder_query_latency), (gst_adder_query):
11566 Implement latency query.
11568 2008-05-27 18:10:00 +0000 Sebastian Dröge <slomo@circular-chaos.org>
11570 gst/adder/gstadder.c: Correctly resync the iterator if gst_iterator_next() returns
11571 Original commit message from CVS:
11572 * gst/adder/gstadder.c: (gst_adder_query_duration):
11573 Correctly resync the iterator if gst_iterator_next() returns
11574 GST_ITERATOR_RESYNC.
11576 2008-05-27 17:14:07 +0000 Tim-Philipp Müller <tim@centricular.net>
11578 win32/vs6/libgstpbutils.dsp: Add pbutils-enumtypes.c to sources (#518037).
11579 Original commit message from CVS:
11580 * win32/vs6/libgstpbutils.dsp:
11581 Add pbutils-enumtypes.c to sources (#518037).
11583 2008-05-27 16:20:17 +0000 Wim Taymans <wim.taymans@gmail.com>
11585 gst-libs/gst/audio/gstaudioclock.*: Add method to inform the clock that the time starts from 0 again. We use this inf...
11586 Original commit message from CVS:
11587 * gst-libs/gst/audio/gstaudioclock.c: (gst_audio_clock_init),
11588 (gst_audio_clock_reset), (gst_audio_clock_get_internal_time):
11589 * gst-libs/gst/audio/gstaudioclock.h:
11590 Add method to inform the clock that the time starts from 0 again. We use
11591 this info to calculate a clock offset so that the time we report in
11592 internal_time is monotonically increasing, as required by the clock base
11593 class. Fixes #521761.
11594 API: GstAudioClock::gst_audio_clock_reset()
11595 * gst-libs/gst/audio/gstbaseaudiosink.c:
11596 (gst_base_audio_sink_skew_slaving),
11597 (gst_base_audio_sink_change_state):
11598 * gst-libs/gst/audio/gstbaseaudiosrc.c:
11599 (gst_base_audio_src_create), (gst_base_audio_src_change_state):
11600 Reset reported time when we (re)create the ringbuffer.
11602 2008-05-27 16:11:32 +0000 Tim-Philipp Müller <tim@centricular.net>
11604 ext/alsa/gstalsamixertrack.c: Make sure playback volumes aren't accidentally overwritten by capture volumes if an als...
11605 Original commit message from CVS:
11606 * ext/alsa/gstalsamixertrack.c:
11607 (gst_alsa_mixer_track_update_alsa_capabilities):
11608 Make sure playback volumes aren't accidentally overwritten by
11609 capture volumes if an alsa mixer track has both playback and
11610 capture capabilities: we create two GstMixerTracks in that
11611 case, so make sure we query only the alsa capabilities that
11612 refer to the type of GstMixerTrack we created from the dual
11613 capability alsa element. Should fix issues with Audigy2 sound
11616 2008-05-27 10:57:56 +0000 Tim-Philipp Müller <tim@centricular.net>
11618 tests/check/pipelines/oggmux.c: Don't use deprecated function.
11619 Original commit message from CVS:
11620 * tests/check/pipelines/oggmux.c: (test_pipeline):
11621 Don't use deprecated function.
11623 2008-05-27 10:35:55 +0000 Wim Taymans <wim.taymans@gmail.com>
11625 gst/playback/gstdecodebin2.c: Check for NULL cases and log them, creating ghostpads can, for example, fail when the p...
11626 Original commit message from CVS:
11627 * gst/playback/gstdecodebin2.c:
11628 (gst_decode_group_control_source_pad), (gst_decode_group_expose):
11629 Check for NULL cases and log them, creating ghostpads can, for example,
11630 fail when the pad returns wrong caps.
11631 * gst/playback/gstplaybin2.c: (perform_eos):
11632 When pushing out the EOS event, collect the return value and warn when
11635 2008-05-26 17:18:52 +0000 Wim Taymans <wim.taymans@gmail.com>
11637 gst-libs/gst/riff/riff-media.c: Add support for DVCPRO.
11638 Original commit message from CVS:
11639 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
11640 (gst_riff_create_video_template_caps):
11641 Add support for DVCPRO.
11643 2008-05-26 10:29:20 +0000 Tim-Philipp Müller <tim@centricular.net>
11645 gst/videoscale/gstvideoscale.c: Change default scaling method from nearest-neighbour to bilinear.
11646 Original commit message from CVS:
11647 * gst/videoscale/gstvideoscale.c: (DEFAULT_PROP_METHOD):
11648 Change default scaling method from nearest-neighbour to bilinear.
11650 2008-05-26 10:26:00 +0000 Tim-Philipp Müller <tim@centricular.net>
11652 tests/check/libs/video.c: More checks.
11653 Original commit message from CVS:
11654 * tests/check/libs/video.c:
11657 2008-05-25 20:51:35 +0000 Tim-Philipp Müller <tim@centricular.net>
11659 Limit duration to a maximum of five seconds for tmplayer format where we can guess the duration only from the timesta...
11660 Original commit message from CVS:
11661 * gst/subparse/gstsubparse.c: (parser_state_init),
11662 (gst_sub_parse_format_autodetect), (handle_buffer):
11663 * gst/subparse/gstsubparse.h:
11664 * tests/check/elements/subparse.c: (test_tmplayer_style3b):
11665 Limit duration to a maximum of five seconds for tmplayer format where
11666 we can guess the duration only from the timestamp of the next line of
11667 text. We don't want to show a text for eternities just because nothing
11668 else is being said for a while.
11670 2008-05-23 14:14:28 +0000 Wim Taymans <wim.taymans@gmail.com>
11672 gst-libs/gst/rtp/gstbasertpdepayload.c: Check sequence numbers, mark input buffers with a discont flag for the subcla...
11673 Original commit message from CVS:
11674 * gst-libs/gst/rtp/gstbasertpdepayload.c:
11675 (gst_base_rtp_depayload_chain),
11676 (gst_base_rtp_depayload_handle_sink_event),
11677 (gst_base_rtp_depayload_push_full),
11678 (gst_base_rtp_depayload_change_state):
11679 Check sequence numbers, mark input buffers with a discont flag for the
11680 subclass when we detected a gap, drop duplicate buffers. We do this
11681 because one can use the element without a jitterbuffer in front and we
11682 don't want to feed the subclasses invalid or reordered data.
11683 Do an error when the subclass did not provide a process function instead
11685 Some other small cleanups.
11687 2008-05-22 22:35:40 +0000 Tim-Philipp Müller <tim@centricular.net>
11689 gst/videotestsrc/videotestsrc.c: May just as well use the precalculated uvstride here.
11690 Original commit message from CVS:
11691 * gst/videotestsrc/videotestsrc.c: (paint_hline_NV12_NV21):
11692 May just as well use the precalculated uvstride here.
11694 2008-05-22 22:09:16 +0000 Jan Schmidt <thaytan@mad.scientist.com>
11696 Add some documentation comments, and some new headers to be scanned.
11697 Original commit message from CVS:
11698 * docs/plugins/Makefile.am:
11699 * docs/plugins/gst-plugins-base-plugins-overrides.txt:
11700 * docs/plugins/gst-plugins-base-plugins-sections.txt:
11701 * docs/plugins/gst-plugins-base-plugins.args:
11702 * docs/plugins/gst-plugins-base-plugins.hierarchy:
11703 * docs/plugins/gst-plugins-base-plugins.interfaces:
11704 * docs/plugins/gst-plugins-base-plugins.prerequisites:
11705 * docs/plugins/inspect/plugin-adder.xml:
11706 * docs/plugins/inspect/plugin-alsa.xml:
11707 * docs/plugins/inspect/plugin-audioconvert.xml:
11708 * docs/plugins/inspect/plugin-audiorate.xml:
11709 * docs/plugins/inspect/plugin-audioresample.xml:
11710 * docs/plugins/inspect/plugin-audiotestsrc.xml:
11711 * docs/plugins/inspect/plugin-cdparanoia.xml:
11712 * docs/plugins/inspect/plugin-decodebin.xml:
11713 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
11714 * docs/plugins/inspect/plugin-gdp.xml:
11715 * docs/plugins/inspect/plugin-gio.xml:
11716 * docs/plugins/inspect/plugin-gnomevfs.xml:
11717 * docs/plugins/inspect/plugin-libvisual.xml:
11718 * docs/plugins/inspect/plugin-ogg.xml:
11719 * docs/plugins/inspect/plugin-pango.xml:
11720 * docs/plugins/inspect/plugin-playback.xml:
11721 * docs/plugins/inspect/plugin-queue2.xml:
11722 * docs/plugins/inspect/plugin-subparse.xml:
11723 * docs/plugins/inspect/plugin-tcp.xml:
11724 * docs/plugins/inspect/plugin-theora.xml:
11725 * docs/plugins/inspect/plugin-typefindfunctions.xml:
11726 * docs/plugins/inspect/plugin-uridecodebin.xml:
11727 * docs/plugins/inspect/plugin-video4linux.xml:
11728 * docs/plugins/inspect/plugin-videorate.xml:
11729 * docs/plugins/inspect/plugin-videoscale.xml:
11730 * docs/plugins/inspect/plugin-videotestsrc.xml:
11731 * docs/plugins/inspect/plugin-volume.xml:
11732 * docs/plugins/inspect/plugin-vorbis.xml:
11733 * docs/plugins/inspect/plugin-ximagesink.xml:
11734 * docs/plugins/inspect/plugin-xvimagesink.xml:
11735 * ext/cdparanoia/gstcdparanoiasrc.c:
11736 * ext/ogg/gstoggdemux.c:
11737 * ext/ogg/gstoggdemux.h:
11738 * ext/ogg/gstoggmux.c:
11739 * ext/ogg/gstoggmux.h:
11740 * gst/audioconvert/audioconvert.c:
11741 * gst/audioconvert/audioconvert.h:
11742 * gst/audioconvert/gstaudioconvert.h:
11743 * gst/gdp/gstgdpdepay.h:
11744 * gst/gdp/gstgdppay.h:
11745 * gst/playback/gstdecodebin.c:
11746 * gst/playback/gstdecodebin2.c:
11747 * gst/playback/gstplaybin.c:
11748 * gst/playback/gstplaybin2.c:
11749 * gst/playback/gsturidecodebin.c:
11750 * gst/tcp/gstmultifdsink.c:
11751 * gst/tcp/gstmultifdsink.h:
11752 * gst/tcp/gsttcp.h:
11753 Add some documentation comments, and some new headers to be scanned.
11754 Rename some internal enum declarations (audioconvert's DitherType and
11755 NoiseShapingType, GstUnitType from the TCP elements) to match the
11756 documented GObject type names so that the docs pick them up.
11757 Name the playbin2 docs markups properly so they get picked up. They'll
11758 need renaming back when/if playbin2 becomes playbin.
11759 100% symbol coverage for the plugin docs, booya.
11761 2008-05-22 18:30:15 +0000 Thijs Vermeir <thijsvermeir@gmail.com>
11763 gst/videotestsrc/videotestsrc.c: Fix generation of NV12/NV21 frames. Fixes bug #532454.
11764 Original commit message from CVS:
11765 Patch by: Thijs Vermeir <thijsvermeir@gmail.com>
11766 * gst/videotestsrc/videotestsrc.c: (paint_hline_NV12_NV21):
11767 Fix generation of NV12/NV21 frames. Fixes bug #532454.
11769 2008-05-22 11:59:33 +0000 Sjoerd Simons <sjoerd@luon.net>
11771 gst/playback/gstdecodebin.c: Lock the fakesink before setting the state to NULL and removing it from the bin so that ...
11772 Original commit message from CVS:
11773 Patch by: Sjoerd Simons <sjoerd at luon dot net>
11774 * gst/playback/gstdecodebin.c: (remove_fakesink):
11775 Lock the fakesink before setting the state to NULL and removing it from
11776 the bin so that a concurrent state change cannot interfere.
11779 2008-05-21 17:09:42 +0000 Felipe Contreras <felipe.contreras@nokia.com>
11781 docs/Makefile.am: Fix installing plugin documentation when gtk-doc is disabled.
11782 Original commit message from CVS:
11783 * docs/Makefile.am:
11784 Fix installing plugin documentation when gtk-doc is disabled.
11786 2008-05-21 17:01:16 +0000 Felipe Contreras <felipe.contreras@nokia.com>
11788 gst-libs/gst/rtsp/Makefile.am: Distribute, don't install md5.h
11789 Original commit message from CVS:
11790 * gst-libs/gst/rtsp/Makefile.am:
11791 Distribute, don't install md5.h
11793 2008-05-21 16:47:58 +0000 Julien Moutte <julien@moutte.net>
11795 gst/tcp/gstmultifdsink.c: Use IPPROTO_IP instead of SOL_IP, works on more platforms.
11796 Original commit message from CVS:
11797 2008-05-21 Julien Moutte <julien@fluendo.com>
11798 * gst/tcp/gstmultifdsink.c: (setup_dscp_client): Use IPPROTO_IP
11799 instead of SOL_IP, works on more platforms.
11800 * gst/typefind/gsttypefindfunctions.c: (aac_type_find): Fix printf
11803 2008-05-21 16:44:15 +0000 Wim Taymans <wim.taymans@gmail.com>
11805 Some debug and comment fixes.
11806 Original commit message from CVS:
11807 * ext/vorbis/vorbisdec.c:
11808 * gst/videoscale/gstvideoscale.c: (gst_video_scale_transform):
11809 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_show_frame):
11810 Some debug and comment fixes.
11811 * tests/examples/dynamic/addstream.c: (main):
11814 2008-05-21 16:36:50 +0000 Wim Taymans <wim.taymans@gmail.com>
11816 Don't use bad gst_element_get_pad().
11817 Original commit message from CVS:
11818 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_typefind):
11819 * gst/playback/decodetest.c: (new_decoded_pad_cb):
11820 * gst/playback/gstdecodebin.c: (gst_decode_bin_init),
11821 (try_to_link_1), (elem_is_dynamic), (close_link), (type_found),
11822 (cleanup_decodebin):
11823 * gst/playback/gstdecodebin2.c: (gst_decode_bin_init),
11824 (connect_element), (gst_decode_group_control_demuxer_pad):
11825 * gst/playback/gstplaybasebin.c: (queue_remove_probe),
11826 (queue_out_of_data), (gen_preroll_element), (preroll_unlinked),
11828 * gst/playback/gstplaybin.c: (gst_play_bin_vis_blocked),
11829 (gst_play_bin_set_property), (handoff), (gen_video_element),
11830 (gen_text_element), (gen_audio_element), (gen_vis_element),
11831 (remove_sinks), (add_sink), (setup_sinks):
11832 * gst/playback/gstplaybin2.c: (pad_added_cb), (no_more_pads_cb):
11833 * gst/playback/gstplaysink.c: (gst_play_sink_get_video_sink),
11834 (gst_play_sink_get_audio_sink), (gst_play_sink_vis_unblocked),
11835 (gst_play_sink_vis_blocked), (gst_play_sink_set_vis_plugin),
11836 (gst_play_sink_get_vis_plugin), (gst_play_sink_set_mute),
11837 (gen_video_chain), (gen_text_chain), (gen_audio_chain),
11838 (gen_vis_chain), (gst_play_sink_reconfigure),
11839 (gst_play_sink_set_font_desc), (gst_play_sink_get_font_desc),
11840 (gst_play_sink_request_pad):
11841 * gst/playback/gsturidecodebin.c: (type_found), (setup_source):
11842 * gst/playback/test.c: (gen_video_element), (gen_audio_element),
11844 * gst/playback/test6.c: (new_decoded_pad_cb):
11845 * tests/check/elements/audioconvert.c: (GST_START_TEST):
11846 * tests/check/elements/audiorate.c: (test_injector_chain),
11847 (do_perfect_stream_test):
11848 * tests/check/elements/ffmpegcolorspace.c: (GST_START_TEST):
11849 * tests/check/elements/gdpdepay.c: (GST_START_TEST):
11850 * tests/check/elements/gnomevfssink.c:
11851 * tests/check/elements/textoverlay.c:
11852 (notgst_check_setup_src_pad2), (notgst_check_teardown_src_pad2):
11853 * tests/check/elements/videotestsrc.c: (GST_START_TEST):
11854 * tests/check/libs/cddabasesrc.c: (GST_START_TEST):
11855 * tests/check/pipelines/oggmux.c: (test_pipeline):
11856 * tests/check/pipelines/streamheader.c: (GST_START_TEST):
11857 * tests/check/pipelines/theoraenc.c: (GST_START_TEST):
11858 * tests/check/pipelines/vorbisenc.c: (GST_START_TEST):
11859 * tests/examples/seek/scrubby.c: (make_wav_pipeline):
11860 * tests/examples/seek/seek.c: (make_mod_pipeline),
11861 (make_dv_pipeline), (make_wav_pipeline), (make_flac_pipeline),
11862 (make_sid_pipeline), (make_parse_pipeline), (make_vorbis_pipeline),
11863 (make_theora_pipeline), (make_vorbis_theora_pipeline),
11864 (make_avi_msmpeg4v3_mp3_pipeline), (make_mp3_pipeline),
11865 (make_avi_pipeline), (make_mpeg_pipeline), (make_mpegnt_pipeline),
11866 (update_fill), (msg_buffering):
11867 Don't use bad gst_element_get_pad().
11869 2008-05-21 14:35:41 +0000 Stefan Kost <ensonic@users.sourceforge.net>
11871 gst-libs/gst/riff/riff-media.c: Fix wrong method name in docs. Fix calculation of strf fields for broken mulaw/alaw.
11872 Original commit message from CVS:
11873 * gst-libs/gst/riff/riff-media.c:
11874 Fix wrong method name in docs. Fix calculation of strf fields for
11876 * gst-libs/gst/riff/riff-read.c:
11877 Whitespace fix and removing double ';'.
11879 2008-05-21 11:52:30 +0000 Wim Taymans <wim.taymans@gmail.com>
11881 docs/design/part-playbin2.txt: Add some leftover doc.
11882 Original commit message from CVS:
11883 * docs/design/part-playbin2.txt:
11884 Add some leftover doc.
11886 2008-05-21 11:36:37 +0000 Sebastian Dröge <slomo@circular-chaos.org>
11888 gst/audioconvert/gstchannelmix.c: Fix copy & paste error in last commit.
11889 Original commit message from CVS:
11890 * gst/audioconvert/gstchannelmix.c: (gst_channel_mix_fill_others):
11891 Fix copy & paste error in last commit.
11893 2008-05-21 11:30:58 +0000 Sebastian Dröge <slomo@circular-chaos.org>
11895 gst/audioconvert/gstchannelmix.c: Add support for mixing GST_AUDIO_CHANNEL_POSITION_SIDE_* from/to other channel posi...
11896 Original commit message from CVS:
11897 * gst/audioconvert/gstchannelmix.c: (gst_channel_mix_fill_others):
11898 Add support for mixing GST_AUDIO_CHANNEL_POSITION_SIDE_* from/to
11899 other channel positions when source has SIDE channels and dest doesn't
11900 or the other way around.
11902 2008-05-21 11:29:25 +0000 Henrik Eriksson <henriken@axis.com>
11904 gst/tcp/gstmultifdsink.*: Add support for DSCP QOS. Fixes #469933.
11905 Original commit message from CVS:
11906 Patch by: Henrik Eriksson <henriken at axis dot com>
11907 * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init),
11908 (gst_multi_fd_sink_init), (setup_dscp_client), (setup_dscp),
11909 (gst_multi_fd_sink_add_full), (gst_multi_fd_sink_set_property),
11910 (gst_multi_fd_sink_get_property):
11911 * gst/tcp/gstmultifdsink.h:
11912 Add support for DSCP QOS. Fixes #469933.
11914 2008-05-21 07:46:02 +0000 Sebastian Dröge <slomo@circular-chaos.org>
11916 tests/check/elements/audioconvert.c: Add another test that checks if conversion between standard 1 and 2 channel layo...
11917 Original commit message from CVS:
11918 * tests/check/elements/audioconvert.c: (GST_START_TEST):
11919 Add another test that checks if conversion between standard 1 and 2
11920 channel layouts with and without positions set is working.
11922 2008-05-21 07:39:56 +0000 Sebastian Dröge <slomo@circular-chaos.org>
11924 gst-libs/gst/audio/multichannel.c: Allow non-standard 2 channel layouts.
11925 Original commit message from CVS:
11926 * gst-libs/gst/audio/multichannel.c:
11927 (gst_audio_check_channel_positions):
11928 Allow non-standard 2 channel layouts.
11929 * tests/check/elements/audioconvert.c: (GST_START_TEST):
11930 Add some tests for converting and remapping non-standard 1 and 2
11933 2008-05-21 07:28:04 +0000 Sebastian Dröge <slomo@circular-chaos.org>
11935 gst/audioconvert/gstchannelmix.c: Prevent division by zero if the channel mix matrix contains only zeroes.
11936 Original commit message from CVS:
11937 * gst/audioconvert/gstchannelmix.c:
11938 (gst_channel_mix_fill_normalize):
11939 Prevent division by zero if the channel mix matrix contains only
11942 2008-05-21 06:45:22 +0000 Antoine Tremblay <hexa00@gmail.com>
11944 gst/gdp/gstgdppay.c: Close a buffer memory leak. Fixes bug #534071.
11945 Original commit message from CVS:
11946 Patch by: Antoine Tremblay <hexa00 at gmail dot com>
11947 * gst/gdp/gstgdppay.c: (gst_gdp_pay_chain):
11948 Close a buffer memory leak. Fixes bug #534071.
11950 2008-05-21 06:39:20 +0000 Sebastian Dröge <slomo@circular-chaos.org>
11952 gst-libs/gst/rtsp/gstrtsptransport.h: Make the GstRTSPTransport struct members public as there are no setters/getters...
11953 Original commit message from CVS:
11954 * gst-libs/gst/rtsp/gstrtsptransport.h:
11955 Make the GstRTSPTransport struct members public as there are no
11956 setters/getters and it's supposed to be changed directly.
11959 2008-05-21 05:48:05 +0000 Sebastian Dröge <slomo@circular-chaos.org>
11961 gst/adder/gstadder.c: Adder also doesn't support audio/x-raw-int with width!=depth so don't claim this on the pad tem...
11962 Original commit message from CVS:
11963 * gst/adder/gstadder.c:
11964 Adder also doesn't support audio/x-raw-int with width!=depth so don't
11965 claim this on the pad template caps.
11967 2008-05-20 16:26:53 +0000 Wim Taymans <wim.taymans@gmail.com>
11969 gst-libs/gst/audio/gstbaseaudiosink.c: We can only use our optimal calibration if we prerolled before the latency exp...
11970 Original commit message from CVS:
11971 * gst-libs/gst/audio/gstbaseaudiosink.c:
11972 (gst_base_audio_sink_sync_latency):
11973 We can only use our optimal calibration if we prerolled before the
11976 2008-05-20 14:35:42 +0000 Tim-Philipp Müller <tim@centricular.net>
11978 configure.ac: Require core CVS for GstBaseSrc buffer caps setting magic.
11979 Original commit message from CVS:
11981 Require core CVS for GstBaseSrc buffer caps setting magic.
11983 2008-05-20 12:26:32 +0000 Sebastian Dröge <slomo@circular-chaos.org>
11985 gst/audioconvert/gstaudioconvert.c: Fix logic in last commit.
11986 Original commit message from CVS:
11987 * gst/audioconvert/gstaudioconvert.c:
11988 (gst_audio_convert_fixate_channels):
11989 Fix logic in last commit.
11991 2008-05-20 12:15:34 +0000 Sebastian Dröge <slomo@circular-chaos.org>
11993 gst/audioconvert/gstaudioconvert.c: Passthrough the channel positions if the number of output channels is the same as...
11994 Original commit message from CVS:
11995 * gst/audioconvert/gstaudioconvert.c:
11996 (gst_audio_convert_fixate_channels):
11997 Passthrough the channel positions if the number of output channels is
11998 the same as the number of input channels, the input had a channel
11999 layout and downstream requests no special one. We did this already for
12000 > 2 channels but now it's also done for 1 channel. Fixes bug #533617.
12002 2008-05-20 11:13:27 +0000 Wim Taymans <wim.taymans@gmail.com>
12004 ext/gnomevfs/gstgnomevfssrc.*: Set the ICY caps on the srcpad from where they get picked up by the base class now and...
12005 Original commit message from CVS:
12006 * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_init),
12007 (gst_gnome_vfs_src_finalize),
12008 (gst_gnome_vfs_src_received_headers_callback),
12009 (gst_gnome_vfs_src_create), (gst_gnome_vfs_src_stop):
12010 * ext/gnomevfs/gstgnomevfssrc.h:
12011 Set the ICY caps on the srcpad from where they get picked up by the base
12012 class now and set on the outgoing buffers.
12013 * gst-libs/gst/audio/gstbaseaudiosrc.c:
12014 (gst_base_audio_src_create):
12015 * sys/v4l/v4lsrc_calls.c: (gst_v4lsrc_buffer_new):
12016 BaseSrc now sets the caps on outgoing buffers automatically.
12018 2008-05-20 11:09:06 +0000 Wim Taymans <wim.taymans@gmail.com>
12020 gst-libs/gst/audio/gstbaseaudiosink.c: Change the way in which the ringbuffer is started when dealing with a slaved c...
12021 Original commit message from CVS:
12022 * gst-libs/gst/audio/gstbaseaudiosink.c:
12023 (gst_base_audio_sink_resample_slaving),
12024 (gst_base_audio_sink_skew_slaving),
12025 (gst_base_audio_sink_sync_latency), (gst_base_audio_sink_render),
12026 (gst_base_audio_sink_async_play),
12027 (gst_base_audio_sink_change_state):
12028 Change the way in which the ringbuffer is started when dealing with a
12029 slaved clock and latency. We now sync to the clock until we reach
12030 upstream latency before starting the ringbuffer. This has the effect
12031 that we can accurately align the master and slave clocks and let the
12032 rate correction code take care of the initial drift or rounding errors
12033 instead of leaving them uncorrected with the old approach.
12035 2008-05-20 08:12:19 +0000 Sebastian Dröge <slomo@circular-chaos.org>
12037 gst/audioconvert/gstaudioconvert.c: Correctly set the default channel positions when converting to 8 channels.
12038 Original commit message from CVS:
12039 * gst/audioconvert/gstaudioconvert.c:
12040 (gst_audio_convert_fixate_channels):
12041 Correctly set the default channel positions when converting to 8
12044 2008-05-19 16:13:25 +0000 Tim-Philipp Müller <tim@centricular.net>
12046 configure.ac: Error out if we don't have the required version of core.
12047 Original commit message from CVS:
12049 Error out if we don't have the required version of core.
12051 2008-05-19 15:59:40 +0000 Tim-Philipp Müller <tim@centricular.net>
12053 gst/typefind/gsttypefindfunctions.c: Use data scan helper in aac typefinder and stop scanning for headers when we've ...
12054 Original commit message from CVS:
12055 * gst/typefind/gsttypefindfunctions.c: (aac_type_find):
12056 Use data scan helper in aac typefinder and stop scanning
12057 for headers when we've found a type. Also fix potential invalid
12058 memory access when calculating the frame length.
12060 2008-05-19 14:09:08 +0000 Tim-Philipp Müller <tim@centricular.net>
12062 gst/typefind/gsttypefindfunctions.c: Don't modify scan context when we return FALSE in ensure_data, so it's possible ...
12063 Original commit message from CVS:
12064 * gst/typefind/gsttypefindfunctions.c: (data_scan_ctx_ensure_data),
12065 (mpeg_sys_is_valid_pack):
12066 Don't modify scan context when we return FALSE in ensure_data, so
12067 it's possible to continue scanning, and we don't end up with a NULL
12068 data pointer and a positive size, which might bite us the next time
12069 we're called. Small constification.
12071 2008-05-16 21:12:02 +0000 Sebastian Dröge <slomo@circular-chaos.org>
12073 gst/adder/gstadder.c: Adder doesn't support 24 bit samples so don't claim it supports them in the pad template caps.
12074 Original commit message from CVS:
12075 * gst/adder/gstadder.c:
12076 Adder doesn't support 24 bit samples so don't claim it supports them
12077 in the pad template caps.
12079 2008-05-14 20:28:02 +0000 Wim Taymans <wim.taymans@gmail.com>
12081 gst-libs/gst/rtp/gstbasertpdepayload.c: Validate the RTP packet before further processing it. It's just too dangerous...
12082 Original commit message from CVS:
12083 * gst-libs/gst/rtp/gstbasertpdepayload.c:
12084 (gst_base_rtp_depayload_chain):
12085 Validate the RTP packet before further processing it. It's just too
12086 dangerous to accept random packets and people are not forced to use a
12087 jitterbuffer or session manager to filter out the bad packets.
12088 * gst-libs/gst/rtp/gstrtpbuffer.c:
12089 (gst_rtp_buffer_set_extension_data),
12090 (gst_rtp_buffer_get_payload_subbuffer):
12092 When setting extension data in a buffer that is too small, we fail and
12093 we should not set the extension bit.
12094 Change GST_WARNINGS into g_warning because they really are
12095 programming errors.
12096 * tests/check/libs/rtp.c: (GST_START_TEST):
12097 Catch the g_warnings now in the unit tests and that fact that failing to
12098 set extension data left the extension bit untouched.
12100 2008-05-14 13:57:41 +0000 Tim-Philipp Müller <tim@centricular.net>
12102 gst/audioresample/gstaudioresample.c: Revert previous change which made basetransform handle buffer_alloc and which b...
12103 Original commit message from CVS:
12104 * gst/audioresample/gstaudioresample.c: (gst_audioresample_init):
12105 Revert previous change which made basetransform handle buffer_alloc
12106 and which breaks things badly in the non-passthrough case since it
12107 returned buffers with a different (ie. sometimes smaller) size than
12108 the size requested.
12110 2008-05-14 13:43:12 +0000 Bernard B <b-gnome@largestprime.net>
12112 gst-libs/gst/rtp/gstrtpbuffer.c: Fix seqnum compare function for bordercase values and fix the docs again. Fixes #533...
12113 Original commit message from CVS:
12114 Patch by: Bernard B <b-gnome at largestprime dot net>
12115 * gst-libs/gst/rtp/gstrtpbuffer.c: (gst_rtp_buffer_compare_seqnum):
12116 Fix seqnum compare function for bordercase values and fix the docs
12117 again. Fixes #533075.
12118 * tests/check/libs/rtp.c: (GST_START_TEST), (rtp_suite):
12119 Add a testcase for seqnum compare function.
12121 2008-05-14 10:58:52 +0000 Sebastian Dröge <slomo@circular-chaos.org>
12123 gst/adder/gstadder.c: Correctly declare the supported endianness on the pad templates and check for correct endiannes...
12124 Original commit message from CVS:
12125 * gst/adder/gstadder.c: (gst_adder_setcaps),
12126 (gst_adder_class_init):
12127 Correctly declare the supported endianness on the pad templates
12128 and check for correct endianness in the set caps function. Adder
12129 only supports native endianness.
12130 Also use gst_element_class_set_details_simple().
12132 2008-05-14 09:12:10 +0000 Stefan Kost <ensonic@users.sourceforge.net>
12134 sys/xvimage/xvimagesink.c: Better debug logging in port value handling. Merging separate port value loops into one.
12135 Original commit message from CVS:
12136 * sys/xvimage/xvimagesink.c:
12137 Better debug logging in port value handling. Merging separate port
12138 value loops into one.
12140 2008-05-13 16:02:19 +0000 Hannes Bistry <hannesb@gmx.de>
12142 gst/tcp/: Fix regression in clientsrc because we did not add the fd to the poll set anymore. Fixes #532364.
12143 Original commit message from CVS:
12144 Patch by: Hannes Bistry <hannesb at gmx dot de>
12145 * gst/tcp/gsttcpclientsrc.c: (gst_tcp_client_src_start):
12146 * gst/tcp/gsttcpserversink.c:
12147 (gst_tcp_server_sink_handle_server_read),
12148 (gst_tcp_server_sink_handle_wait), (gst_tcp_server_sink_init_send):
12149 Fix regression in clientsrc because we did not add the fd to the poll
12150 set anymore. Fixes #532364.
12151 Do some cleanups here and there.
12153 2008-05-13 13:04:24 +0000 Sebastian Dröge <slomo@circular-chaos.org>
12155 gst/playback/: Use correct marshallers. GstCaps are a boxed type and no GObject subclass.
12156 Original commit message from CVS:
12157 * gst/playback/gstdecodebin.c: (gst_decode_bin_class_init):
12158 * gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init):
12159 * gst/playback/gstplay-marshal.list:
12160 * gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_class_init):
12161 Use correct marshallers. GstCaps are a boxed type and no GObject
12164 2008-05-13 11:37:15 +0000 Sebastian Dröge <slomo@circular-chaos.org>
12166 win32/common/libgstrtsp.def: Add gst_rtsp_connection_(set|clear)_auth_param() to the exported symbols.
12167 Original commit message from CVS:
12168 * win32/common/libgstrtsp.def:
12169 Add gst_rtsp_connection_(set|clear)_auth_param() to the exported
12172 2008-05-13 10:59:49 +0000 Sjoerd Simons <sjoerd@luon.net>
12174 tests/check/elements/audioresample.c: Add unit test for the latest basetransform negotiation changes.
12175 Original commit message from CVS:
12176 Patch by: Sjoerd Simons <sjoerd at luon dot net>
12177 * tests/check/elements/audioresample.c:
12178 (live_switch_alloc_only_48000), (live_switch_get_sink_caps),
12179 (live_switch_push), (GST_START_TEST):
12180 Add unit test for the latest basetransform negotiation changes.
12183 2008-05-13 09:14:44 +0000 Sebastian Dröge <slomo@circular-chaos.org>
12185 gst/ffmpegcolorspace/imgconvert.c: Fix nv12<->nv21 conversion if stride is larger than width.
12186 Original commit message from CVS:
12187 * gst/ffmpegcolorspace/imgconvert.c: (nv12_to_nv21):
12188 Fix nv12<->nv21 conversion if stride is larger than width.
12190 2008-05-13 07:28:21 +0000 j^ <j@oil21.org>
12192 ext/ogg/gstoggdemux.*: Parse presentation time from skeleton streams and use it as offset for the timestamps. Fixes b...
12193 Original commit message from CVS:
12194 Patch by: j^ <j at oil21 dot org>
12195 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_parse_skeleton_fishead),
12196 (gst_ogg_pad_parse_skeleton_fisbone):
12197 * ext/ogg/gstoggdemux.h:
12198 Parse presentation time from skeleton streams and use it as offset
12199 for the timestamps. Fixes bug #530068.
12201 2008-05-12 08:45:11 +0000 Wim Taymans <wim.taymans@gmail.com>
12203 gst-libs/gst/audio/gstbaseaudiosink.c: Revert previous patch that attempted to more accurately calculate the initial ...
12204 Original commit message from CVS:
12205 * gst-libs/gst/audio/gstbaseaudiosink.c:
12206 (gst_base_audio_sink_render), (gst_base_audio_sink_async_play):
12207 Revert previous patch that attempted to more accurately calculate the
12208 initial offset between master and slave clock. The best thing we can do
12209 in general is take the time of both clocks as the diff since we don't
12210 know when the actual preroll happened.
12212 2008-05-11 19:52:59 +0000 Tim-Philipp Müller <tim@centricular.net>
12214 gst-libs/gst/pbutils/install-plugins.c: Fix docs: type and missing word.
12215 Original commit message from CVS:
12216 * gst-libs/gst/pbutils/install-plugins.c:
12217 Fix docs: type and missing word.
12219 2008-05-10 20:16:21 +0000 Tim-Philipp Müller <tim@centricular.net>
12221 gst/typefind/gsttypefindfunctions.c: Don't do lots of 4-byte peeks, but use the 'new' data scan helper for this inste...
12222 Original commit message from CVS:
12223 * gst/typefind/gsttypefindfunctions.c: (h264_video_type_find):
12224 Don't do lots of 4-byte peeks, but use the 'new' data scan helper
12225 for this instead; don't check if we've found enough markers after
12226 each and every step, it's enough to do that only if we've actually
12227 found a new marker.
12228 Embed a G_UNLIKELY into the IS_MPEG_HEADER macro.
12230 2008-05-10 18:19:17 +0000 Tim-Philipp Müller <tim@centricular.net>
12232 gst/typefind/gsttypefindfunctions.c: Move scan helper thingy to the beginning of the file so we can use it in other t...
12233 Original commit message from CVS:
12234 * gst/typefind/gsttypefindfunctions.c:
12235 (DATA_SCAN_CTX_CHUNK_SIZE), (DataScanCtx), (data_scan_ctx_advance),
12236 (data_scan_ctx_ensure_data), (GST_MPEGVID_TYPEFIND_TRY_SYNC),
12237 (mpeg_video_stream_type_find):
12238 Move scan helper thingy to the beginning of the file so we can use
12239 it in other typefind functions. Rename it to something more
12240 generic. Also improve handling of things towards the end of the
12241 typefind data: peek as much as we can if we know the size of the
12242 data, rather than just min_size.
12244 2008-05-09 21:42:26 +0000 Jan Schmidt <thaytan@mad.scientist.com>
12246 Document the GstTuner and GstColorBalance interfaces, and some other random API functions that needed it. 70% symbol ...
12247 Original commit message from CVS:
12248 * docs/libs/gst-plugins-base-libs-sections.txt:
12249 * gst-libs/gst/interfaces/colorbalance.c:
12250 * gst-libs/gst/interfaces/colorbalance.h:
12251 * gst-libs/gst/interfaces/colorbalancechannel.c:
12252 * gst-libs/gst/interfaces/colorbalancechannel.h:
12253 * gst-libs/gst/interfaces/tuner.c:
12254 * gst-libs/gst/interfaces/tunerchannel.c:
12255 * gst-libs/gst/interfaces/tunerchannel.h:
12256 * gst-libs/gst/interfaces/tunernorm.c:
12257 * gst-libs/gst/interfaces/tunernorm.h:
12258 * gst-libs/gst/video/video.c:
12259 * gst-libs/gst/video/video.h:
12260 Document the GstTuner and GstColorBalance interfaces, and some
12261 other random API functions that needed it. 70% symbol coverage, woo.
12263 2008-05-09 16:38:10 +0000 Wim Taymans <wim.taymans@gmail.com>
12265 gst-libs/gst/audio/gstaudiosink.c: Choose to allocate one less segment but require one additional segment as latency.
12266 Original commit message from CVS:
12267 * gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_acquire):
12268 Choose to allocate one less segment but require one additional segment
12270 * gst-libs/gst/audio/gstaudiosrc.c: (gst_audioringbuffer_acquire):
12271 No need to increment the number of segments in the source.
12272 * gst-libs/gst/audio/gstbaseaudiosink.c:
12273 (gst_base_audio_sink_get_time), (clock_convert_external),
12274 (gst_base_audio_sink_resample_slaving),
12275 (gst_base_audio_sink_skew_slaving),
12276 (gst_base_audio_sink_none_slaving), (gst_base_audio_sink_render),
12277 (gst_base_audio_sink_async_play):
12278 Remove adding latency when returning the internal time while subtracting
12279 it again when we use the value a little later.
12280 When calculating the end timestamp, we are making a rounding error
12281 with the current algorithm. Ensure that we don't accumulate these
12282 rounding errors when aligning samples by not resampling at all if we
12283 don't need to. Fixes #419351.
12284 Make the initial calibration of the clock slaving a little more
12285 predictable and accurate. Also handle the case where we don't do
12288 2008-05-09 08:34:52 +0000 Sebastian Dröge <slomo@circular-chaos.org>
12290 gst/ffmpegcolorspace/: Add conversions from/to NV12 and NV21 and conversions between those two formats. Fixes bug #53...
12291 Original commit message from CVS:
12292 Based on a patch by:
12293 Björn Benderius <bjoern dot benderius at axis dot com>
12294 * gst/ffmpegcolorspace/avcodec.h:
12295 * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
12296 (gst_ffmpeg_pixfmt_to_caps), (gst_ffmpeg_caps_to_pixfmt),
12297 (gst_ffmpegcsp_avpicture_fill):
12298 * gst/ffmpegcolorspace/imgconvert.c: (nv12_to_nv21):
12299 * gst/ffmpegcolorspace/imgconvert_template.h:
12300 Add conversions from/to NV12 and NV21 and conversions between those
12301 two formats. Fixes bug #532166.
12303 2008-05-08 17:35:44 +0000 Edward Hervey <bilboed@bilboed.com>
12305 gst/typefind/gsttypefindfunctions.c: Abort the h264 typefinding as soon as _peek() doesn't return anything, which hap...
12306 Original commit message from CVS:
12307 * gst/typefind/gsttypefindfunctions.c: (h264_video_type_find):
12308 Abort the h264 typefinding as soon as _peek() doesn't return anything,
12309 which happens for example with files smaller than 128kb.
12311 2008-05-08 14:46:27 +0000 Wouter Cloetens <zombie@e2big.org>
12313 gst-libs/gst/rtsp/: Add Digest authorization support for RTSP connections. See #532065.
12314 Original commit message from CVS:
12315 Patch by: Wouter Cloetens <zombie at e2big dot org>
12316 * gst-libs/gst/rtsp/Makefile.am:
12317 * gst-libs/gst/rtsp/gstrtspconnection.c:
12318 (gst_rtsp_connection_create), (md5_digest_to_hex_string),
12319 (auth_digest_compute_hex_urp), (auth_digest_compute_response),
12320 (add_auth_header), (gst_rtsp_connection_free),
12321 (gst_rtsp_connection_set_auth), (str_case_hash), (str_case_equal),
12322 (gst_rtsp_connection_set_auth_param),
12323 (gst_rtsp_connection_clear_auth_params):
12324 * gst-libs/gst/rtsp/gstrtspconnection.h:
12325 Add Digest authorization support for RTSP connections. See #532065.
12326 * gst-libs/gst/rtsp/md5.c:
12327 * gst-libs/gst/rtsp/md5.h:
12328 Yeap, another md5 implementation until we can depend on a glib that has
12331 2008-05-08 06:20:42 +0000 Sjoerd Simons <sjoerd@luon.net>
12333 gst/audioresample/gstaudioresample.c: Let audioresample use the buffer allocation of basetransform instead of it's ow...
12334 Original commit message from CVS:
12335 Patch by: Sjoerd Simons <sjoerd at luon dot net>
12336 * gst/audioresample/gstaudioresample.c: (gst_audioresample_init):
12337 Let audioresample use the buffer allocation of basetransform instead
12339 * tests/check/elements/audioresample.c: (alloc_only_48000),
12340 (GST_START_TEST), (audioresample_suite):
12341 Add unit test for the recent basetransform bugfix, where upstream
12342 changes caps to something that can't be passed through anymore.
12344 2008-05-07 19:50:27 +0000 Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>
12346 win32/common/config.h.in: Don't define GST_FUNCTION, if GLib supports MSVC we'd much rather use the real thing than h...
12347 Original commit message from CVS:
12348 * win32/common/config.h.in:
12349 Don't define GST_FUNCTION, if GLib supports MSVC we'd much rather
12350 use the real thing than having "???" unconditionally.
12352 2008-05-07 15:47:03 +0000 Wim Taymans <wim.taymans@gmail.com>
12354 gst-libs/gst/audio/gstbaseaudiosink.c: Report the latency with the new seglatency parameter.
12355 Original commit message from CVS:
12356 * gst-libs/gst/audio/gstbaseaudiosink.c:
12357 (gst_base_audio_sink_query):
12358 Report the latency with the new seglatency parameter.
12359 * gst-libs/gst/audio/gstringbuffer.c:
12360 (gst_ring_buffer_debug_spec_buff), (gst_ring_buffer_parse_caps),
12361 (gst_ring_buffer_acquire):
12362 * gst-libs/gst/audio/gstringbuffer.h:
12363 Add new field to the ringbufferspec to specify the expected latency
12364 between the underlying device read/write pointer, this is needed
12365 when writing sinks that sit a little closer to the hardware.
12366 Add some more docs for other fields.
12368 2008-05-07 10:38:23 +0000 Wim Taymans <wim.taymans@gmail.com>
12370 gst-libs/gst/app/: Add marshal.list, make it compile and add to cvsignore.
12371 Original commit message from CVS:
12372 * gst-libs/gst/app/.cvsignore:
12373 * gst-libs/gst/app/Makefile.am:
12374 * gst-libs/gst/app/gstapp-marshal.list:
12375 Add marshal.list, make it compile and add to cvsignore.
12376 * gst-libs/gst/app/gstappsink.c: (gst_app_sink_dispose),
12377 (gst_app_sink_stop):
12379 * gst-libs/gst/app/gstappsrc.c: (gst_app_src_class_init),
12380 (gst_app_src_init), (gst_app_src_set_property),
12381 (gst_app_src_get_property), (gst_app_src_unlock),
12382 (gst_app_src_unlock_stop), (gst_app_src_start), (gst_app_src_stop),
12383 (gst_app_src_create), (gst_app_src_set_caps),
12384 (gst_app_src_get_caps), (gst_app_src_set_size),
12385 (gst_app_src_get_size), (gst_app_src_set_seekable),
12386 (gst_app_src_get_seekable), (gst_app_src_set_max_buffers),
12387 (gst_app_src_get_max_buffers), (gst_app_src_push_buffer),
12388 (gst_app_src_end_of_stream):
12389 * gst-libs/gst/app/gstappsrc.h:
12390 Beat appsrc in shape, add signals and actions.
12392 Add properties for caps, size, seekability and max-buffers.
12393 Fix unlock/stop code.
12395 2008-05-06 12:35:09 +0000 Sebastian Dröge <slomo@circular-chaos.org>
12397 gst/volume/gstvolume.c: Return NOT_NEGOTIATED if we didn't set a process function yet for some reason instead of cras...
12398 Original commit message from CVS:
12399 * gst/volume/gstvolume.c: (volume_transform_ip):
12400 Return NOT_NEGOTIATED if we didn't set a process function yet for some
12401 reason instead of crashing later. Might fix bug #509125.
12403 2008-05-06 12:12:16 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
12405 gst/audioconvert/: Add support for more than 8 channels and NONE channel layouts. For more than 8 channels no channel...
12406 Original commit message from CVS:
12407 Based on a patch by: Tim-Philipp Müller <tim.muller at collabora co uk>
12408 * gst/audioconvert/audioconvert.c: (audio_convert_prepare_context):
12409 * gst/audioconvert/audioconvert.h:
12410 * gst/audioconvert/gstaudioconvert.c:
12411 (gst_audio_convert_parse_caps),
12412 (structure_has_fixed_channel_positions),
12413 (gst_audio_convert_transform_caps):
12414 * gst/audioconvert/gstchannelmix.c: (gst_channel_mix_fill_matrix):
12415 Add support for more than 8 channels and NONE channel layouts. For
12416 more than 8 channels no channel conversion is supported yet, only
12417 format conversions are supported. Fixes bug #398033.
12418 * tests/check/elements/audioconvert.c: (verify_convert),
12419 (GST_START_TEST), (audioconvert_suite):
12420 Add some unit tests by Tim for checking the NONE channel layouts
12421 and more than 8 channels and add some more unit tests for channel
12424 2008-05-06 10:16:49 +0000 Wim Taymans <wim.taymans@gmail.com>
12426 gst/playback/gstdecodebin2.c: When autoplugging fails, set the element back to NULL before unreffing it.
12427 Original commit message from CVS:
12428 * gst/playback/gstdecodebin2.c: (connect_pad):
12429 When autoplugging fails, set the element back to NULL before
12432 2008-05-06 09:59:43 +0000 Sebastian Dröge <slomo@circular-chaos.org>
12434 win32/common/libgstaudio.def: Add gst_base_audio_src_[sg]et_slave_method() to the exported symbols.
12435 Original commit message from CVS:
12436 * win32/common/libgstaudio.def:
12437 Add gst_base_audio_src_[sg]et_slave_method() to the exported
12440 2008-05-05 12:33:05 +0000 Sebastian Dröge <slomo@circular-chaos.org>
12442 gst/subparse/samiparse.c: Remove trailing, leading and double whitespaces.
12443 Original commit message from CVS:
12444 * gst/subparse/samiparse.c: (handle_start_sync),
12445 (end_sami_element), (characters_sami):
12446 Remove trailing, leading and double whitespaces.
12447 Correctly timestamp buffers and output the last buffer too.
12448 * tests/check/elements/subparse.c: (GST_START_TEST),
12450 Add a simple unit test for SAMI parsing.
12452 2008-05-05 11:14:48 +0000 Young-Ho Cha <ganadist@chollian.net>
12454 gst/subparse/samiparse.c: Only output characters inside the "sync" elements. There could be other elements like "styl...
12455 Original commit message from CVS:
12456 Patch by: Young-Ho Cha <ganadist at chollian dot net>
12457 * gst/subparse/samiparse.c: (handle_start_sync),
12458 (start_sami_element), (end_sami_element), (characters_sami),
12459 (sami_context_reset):
12460 Only output characters inside the "sync" elements. There could be
12461 other elements like "style" that have some content but should
12462 not be printed. Fixes bug #467911.
12464 2008-05-05 10:27:45 +0000 Wim Taymans <wim.taymans@gmail.com>
12466 gst-libs/gst/app/gstappsink.*: Start some docs.
12467 Original commit message from CVS:
12468 * gst-libs/gst/app/gstappsink.c: (gst_app_sink_class_init),
12469 (gst_app_sink_init), (gst_app_sink_set_property),
12470 (gst_app_sink_get_property), (gst_app_sink_unlock_start),
12471 (gst_app_sink_unlock_stop), (gst_app_sink_flush_unlocked),
12472 (gst_app_sink_start), (gst_app_sink_stop), (gst_app_sink_event),
12473 (gst_app_sink_preroll), (gst_app_sink_render),
12474 (gst_app_sink_set_caps), (gst_app_sink_set_drop),
12475 (gst_app_sink_get_drop):
12476 * gst-libs/gst/app/gstappsink.h:
12478 Add property to drop buffers when the queue is filled
12479 Fix unlocking and flushing when the queues are filled.
12481 2008-05-05 10:03:51 +0000 Sebastian Dröge <slomo@circular-chaos.org>
12483 gst/playback/: Allow setting -1 as current-audio to mute the current audio stream, similar to what is done for subtit...
12484 Original commit message from CVS:
12485 * gst/playback/gstplaybasebin.c: (set_audio_mute),
12486 (set_active_source):
12487 * gst/playback/gstplaybasebin.h:
12488 * gst/playback/gstplaybin.c: (gst_play_bin_class_init),
12489 (playbin_set_audio_mute):
12490 Allow setting -1 as current-audio to mute the current audio stream,
12491 similar to what is done for subtitles. Fixes bug #342294.
12493 2008-05-05 07:41:03 +0000 Edward Hervey <bilboed@bilboed.com>
12495 gst-libs/gst/pbutils/descriptions.c: It's SorensOn and not SorensEn.
12496 Original commit message from CVS:
12497 * gst-libs/gst/pbutils/descriptions.c: (formats):
12498 It's SorensOn and not SorensEn.
12500 2008-05-04 15:23:36 +0000 Tim-Philipp Müller <tim@centricular.net>
12502 gst-libs/gst/pbutils/descriptions.c: Fix description of video/x-flash-video.
12503 Original commit message from CVS:
12504 * gst-libs/gst/pbutils/descriptions.c: (formats):
12505 Fix description of video/x-flash-video.
12507 2008-05-04 15:02:20 +0000 Sebastian Dröge <slomo@circular-chaos.org>
12509 Remove some unused code.
12510 Original commit message from CVS:
12511 * gst-libs/gst/audio/gstaudiosink.c: (audioringbuffer_thread_func):
12512 * gst-libs/gst/audio/gstaudiosrc.c: (audioringbuffer_thread_func):
12513 * gst/tcp/gsttcp.c: (gst_tcp_socket_write):
12514 * sys/v4l/v4lsrc_calls.c: (gst_v4lsrc_get_fps_list):
12515 Remove some unused code.
12516 * gst/audioconvert/gstaudioquantize.c:
12517 (gst_audio_quantize_free_noise_shaping):
12518 Don't return before freeing the noise shaping history.
12520 2008-05-03 16:00:04 +0000 Tim-Philipp Müller <tim@centricular.net>
12522 tests/check/elements/subparse.c: Add unit test for the tmplayer variant from bug #530962.
12523 Original commit message from CVS:
12524 * tests/check/elements/subparse.c: (do_test),
12525 (test_tmplayer_style3b), (subparse_suite):
12526 Add unit test for the tmplayer variant from bug #530962.
12528 2008-05-03 15:45:23 +0000 Tim-Philipp Müller <tim@centricular.net>
12530 gst/subparse/: Fix parsing of tmplayer subtitle variant where every single line contains text and there isn't an empt...
12531 Original commit message from CVS:
12532 * gst/subparse/gstsubparse.c: (handle_buffer),
12533 (gst_sub_parse_sink_event):
12534 * gst/subparse/tmplayerparse.c: (tmplayer_process_buffer),
12535 (tmplayer_parse_line):
12536 Fix parsing of tmplayer subtitle variant where every single line contains
12537 text and there isn't an empty line after each line to determine the
12538 duration (#530962). Improve EOS handling for tmplayer subtitles a bit by
12539 making sure that we push out the last line of text without a duration if
12540 there's still text left in the buffer at the end.
12542 2008-05-03 15:39:04 +0000 Tim-Philipp Müller <tim@centricular.net>
12544 gst/subparse/gstsubparse.c: Fix detection of discontinuities based on the buffer offset (doesn't work so well if no b...
12545 Original commit message from CVS:
12546 * gst/subparse/gstsubparse.c: (feed_textbuf):
12547 Fix detection of discontinuities based on the buffer offset (doesn't work
12548 so well if no buffer offset is set) and also check for the DISCONT buffer
12549 flag. This keeps the parser state from being reset after each buffer in
12552 2008-05-03 12:09:16 +0000 Tim-Philipp Müller <tim@centricular.net>
12554 gst/typefind/gsttypefindfunctions.c: Further fine-tuning: don't absolutely require sequence or GOP headers but adjust...
12555 Original commit message from CVS:
12556 * gst/typefind/gsttypefindfunctions.c: (mpeg_video_stream_type_find):
12557 Further fine-tuning: don't absolutely require sequence or GOP headers
12558 (as introduced in the previous commit), but adjust the typefind
12559 probabilities returned accordingly if we don't see them. Also make sure
12560 picture header and first slice are somewhat close to each other (which
12561 is not perfect but still better than requiring a fixed offset or having
12564 2008-05-02 12:13:08 +0000 Wim Taymans <wim.taymans@gmail.com>
12566 gst-libs/gst/rtp/gstbasertppayload.c: Rename the setcaps/getcaps function internally to make it clear that they are c...
12567 Original commit message from CVS:
12568 * gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_init),
12569 (gst_basertppayload_sink_setcaps),
12570 (gst_basertppayload_sink_getcaps):
12571 Rename the setcaps/getcaps function internally to make it clear that
12572 they are called for the sink pad.
12574 2008-05-02 12:11:07 +0000 Wim Taymans <wim.taymans@gmail.com>
12576 gst-libs/gst/rtp/gstbasertpdepayload.*: Catch packet-lost events from the jitterbuffer and convert them into a vmetho...
12577 Original commit message from CVS:
12578 * gst-libs/gst/rtp/gstbasertpdepayload.c:
12579 (gst_base_rtp_depayload_class_init),
12580 (gst_base_rtp_depayload_handle_sink_event), (create_segment_event),
12581 (gst_base_rtp_depayload_packet_lost),
12582 (gst_base_rtp_depayload_set_gst_timestamp):
12583 * gst-libs/gst/rtp/gstbasertpdepayload.h:
12584 Catch packet-lost events from the jitterbuffer and convert them into a
12585 vmethod call (lost-packet) so that depayloaders can do something smart.
12586 Also add a default packet-lost function that sends out a segment update
12589 2008-05-02 11:13:05 +0000 Stefan Kost <ensonic@users.sourceforge.net>
12591 gst/playback/: Also include config.h when relying on defines from it. Fixes the build. Its been a please to serve :)
12592 Original commit message from CVS:
12593 * gst/playback/test4.c:
12594 * gst/playback/test5.c:
12595 * gst/playback/test6.c:
12596 * gst/playback/test7.c:
12597 Also include config.h when relying on defines from it. Fixes the
12598 build. Its been a please to serve :)
12600 2008-05-02 10:54:51 +0000 Thijs Vermeir <thijsvermeir@gmail.com>
12603 * gst/videotestsrc/videotestsrc.c:
12604 Add support for NV12 and NV21 in videotestsrc
12605 Original commit message from CVS:
12606 * gst/videotestsrc/videotestsrc.c (paint_setup_NV12),
12607 (paint_setup_NV21), (paint_hline_NV12_NV21):
12608 Add support for NV12 and NV21 in videotestsrc
12610 2008-05-02 10:02:05 +0000 Sebastian Dröge <slomo@circular-chaos.org>
12612 gst/videoscale/: Support 1x1 images as input and output as for example the BBC HQ new streams have 1x1 GIFs in the pl...
12613 Original commit message from CVS:
12614 * gst/videoscale/gstvideoscale.c:
12615 * gst/videoscale/vs_4tap.c: (vs_image_scale_4tap_Y):
12616 * gst/videoscale/vs_image.c: (vs_image_scale_nearest_RGBA),
12617 (vs_image_scale_linear_RGBA), (vs_image_scale_nearest_RGB),
12618 (vs_image_scale_linear_RGB), (vs_image_scale_nearest_YUYV),
12619 (vs_image_scale_linear_YUYV), (vs_image_scale_nearest_UYVY),
12620 (vs_image_scale_linear_UYVY), (vs_image_scale_nearest_Y),
12621 (vs_image_scale_linear_Y), (vs_image_scale_nearest_RGB565),
12622 (vs_image_scale_linear_RGB565), (vs_image_scale_nearest_RGB555),
12623 (vs_image_scale_linear_RGB555):
12624 Support 1x1 images as input and output as for example the BBC HQ new
12625 streams have 1x1 GIFs in the playlists for some reason.
12627 2008-05-01 19:11:42 +0000 Tim-Philipp Müller <tim@centricular.net>
12629 gst/playback/gstdecodebin.c: If we can't activate one of the decoders we plugged in (such as, say, musepackdec) for s...
12630 Original commit message from CVS:
12631 * gst/playback/gstdecodebin.c: (free_pad_probe_for_element),
12633 If we can't activate one of the decoders we plugged in (such as,
12634 say, musepackdec) for some reason (it might not support push mode,
12635 for example), remove any pad probes that close_pad_link() might
12636 have set up. This makes sure we later don't try to remove a probe
12637 for a pad that doesn't exist any longer, and avoids nast warnings
12638 and probably other things too.
12640 2008-04-30 20:54:56 +0000 Tim-Philipp Müller <tim@centricular.net>
12642 gst/typefind/gsttypefindfunctions.c: Rework mpeg video stream typefinding a bit more: make sure sequence,
12643 Original commit message from CVS:
12644 * gst/typefind/gsttypefindfunctions.c:
12645 (mpeg_video_stream_ctx_ensure_data), (mpeg_video_stream_type_find),
12647 Rework mpeg video stream typefinding a bit more: make sure sequence,
12648 GOP, picture and slice headers appear in the order they should and
12649 that we've in fact at least had one of each; fix picture header
12650 detection; decouple picture and slice header check - don't assume
12651 they're at a fixed offset, there may be extra data in between. Also,
12652 announce varying degrees of probability depending on what we found
12653 exactly (multiple pictures, at least one picture, just sequence and
12654 GOP headers). Finally, in _ensure_data(), take into account that we
12655 might be typefinding smaller amounts of data, such as the first
12656 buffer of a stream, so fall back to the minimum size needed as long
12657 as that's available, instead of erroring out if there's less than
12658 2kB of data. Fixes #526173. Conveniently also doesn't recognise the
12659 fuzzed file from #399342 as valid.
12661 2008-04-30 17:06:45 +0000 Michael Smith <msmith@xiph.org>
12663 ext/theora/theoradec.c: Cool kids don't divide by zero.
12664 Original commit message from CVS:
12665 * ext/theora/theoradec.c:
12666 Cool kids don't divide by zero.
12667 Treat PAR of x:0 as 1:1.
12670 2008-04-30 14:37:52 +0000 Tim-Philipp Müller <tim@centricular.net>
12672 gst/typefind/gsttypefindfunctions.c: Refactor a bit: use context structure to track parsing offset and size of availa...
12673 Original commit message from CVS:
12674 * gst/typefind/gsttypefindfunctions.c: (MpegVideoStreamCtx),
12675 (mpeg_video_stream_ctx_advance), (mpeg_video_stream_ctx_ensure_data),
12676 (mpeg_video_stream_type_find):
12677 Refactor a bit: use context structure to track parsing offset and size of
12678 available data and make the code a bit clearer. Fixes bad memory access
12681 2008-04-28 22:18:49 +0000 Michael Smith <msmith@xiph.org>
12683 gst/: Include stdlib.h and unistd.h only if the appropriate HAVE_*_H macro is defined.
12684 Original commit message from CVS:
12685 * gst/playback/test4.c:
12686 * gst/playback/test5.c:
12687 * gst/playback/test6.c:
12688 * gst/tcp/gstmultifdsink.c:
12689 Include stdlib.h and unistd.h only if the appropriate HAVE_*_H macro
12692 2008-04-28 08:51:38 +0000 Wim Taymans <wim.taymans@gmail.com>
12694 gst-libs/gst/audio/gstbaseaudiosink.h: Clarify some docs.
12695 Original commit message from CVS:
12696 * gst-libs/gst/audio/gstbaseaudiosink.h:
12698 * gst-libs/gst/audio/gstbaseaudiosrc.c: (slave_method_get_type),
12699 (gst_base_audio_src_class_init), (gst_base_audio_src_init),
12700 (gst_base_audio_src_set_slave_method),
12701 (gst_base_audio_src_get_slave_method),
12702 (gst_base_audio_src_set_property),
12703 (gst_base_audio_src_get_property), (gst_base_audio_src_create):
12704 * gst-libs/gst/audio/gstbaseaudiosrc.h:
12705 Add property and methods for selecting the clock slave method in the
12706 source, like in the sink.
12707 We only implement "none" and "re-timestamp" for now.
12708 API: gst_base_audio_src_set_slave_method()
12709 API: gst_base_audio_src_get_slave_method()
12711 2008-04-25 18:18:47 +0000 Wim Taymans <wim.taymans@gmail.com>
12713 gst-libs/gst/app/gstappsink.*: Add more docs.
12714 Original commit message from CVS:
12715 * gst-libs/gst/app/gstappsink.c: (gst_app_sink_class_init),
12716 (gst_app_sink_init), (gst_app_sink_set_property),
12717 (gst_app_sink_get_property), (gst_app_sink_event),
12718 (gst_app_sink_preroll), (gst_app_sink_render),
12719 (gst_app_sink_set_emit_signals), (gst_app_sink_get_emit_signals),
12720 (gst_app_sink_set_max_buffers), (gst_app_sink_get_max_buffers),
12721 (gst_app_sink_pull_buffer):
12722 * gst-libs/gst/app/gstappsink.h:
12724 Add signals for when preroll and render buffers are available.
12725 Add property to control signal emission.
12726 Add property to control the max queue size.
12728 2008-04-25 07:37:09 +0000 Wim Taymans <wim.taymans@gmail.com>
12730 gst-libs/gst/rtp/gstrtpbuffer.c: Fix the docs about the seqnum compare function, it returns a difference.
12731 Original commit message from CVS:
12732 * gst-libs/gst/rtp/gstrtpbuffer.c:
12733 Fix the docs about the seqnum compare function, it returns a difference.
12735 2008-04-24 09:27:35 +0000 Edward Hervey <bilboed@bilboed.com>
12737 ext/alsa/gstalsadeviceprobe.c: Don't return before freeing up the allocated structures.
12738 Original commit message from CVS:
12739 * ext/alsa/gstalsadeviceprobe.c:
12740 (gst_alsa_get_device_list): Don't return before freeing up
12741 the allocated structures.
12743 2008-04-24 08:19:35 +0000 Stefan Kost <ensonic@users.sourceforge.net>
12745 gst/playback/gstplaybin.c: Remove obsolete streaminfo code and fix a leak. Fixes #529546
12746 Original commit message from CVS:
12747 * gst/playback/gstplaybin.c:
12748 Remove obsolete streaminfo code and fix a leak. Fixes #529546
12750 2008-04-23 13:50:34 +0000 Stefan Kost <ensonic@users.sourceforge.net>
12752 ext/ogg/gstoggdemux.c: Revert the event part, that should not go in.
12753 Original commit message from CVS:
12754 * ext/ogg/gstoggdemux.c:
12755 Revert the event part, that should not go in.
12757 2008-04-23 13:45:29 +0000 Stefan Kost <ensonic@users.sourceforge.net>
12759 ext/ogg/gstoggdemux.c: Don't leak GstPluginFeatures when filtering.
12760 Original commit message from CVS:
12761 * ext/ogg/gstoggdemux.c:
12762 Don't leak GstPluginFeatures when filtering.
12764 2008-04-23 08:58:42 +0000 Stefan Kost <ensonic@users.sourceforge.net>
12766 sys/xvimage/xvimagesink.c: Add some logging for cases when grabbing the xv failed.
12767 Original commit message from CVS:
12768 * sys/xvimage/xvimagesink.c:
12769 Add some logging for cases when grabbing the xv failed.
12771 2008-04-22 06:18:04 +0000 David Schleef <ds@schleef.org>
12773 ext/ogg/gstoggmux.c: Update Ogg/Dirac muxing. Removes the weird "KW-DIRAC" bos packet. Should conform to what we cu...
12774 Original commit message from CVS:
12775 * ext/ogg/gstoggmux.c:
12776 Update Ogg/Dirac muxing. Removes the weird "KW-DIRAC" bos
12777 packet. Should conform to what we currently think is the
12778 final Ogg/Dirac muxing spec.
12780 2008-04-22 06:13:43 +0000 David Schleef <ds@schleef.org>
12782 sys/xvimage/xvimagesink.c: Fix typo that causes the overlay keying color to bright green on a 16-bit display. Dark g...
12783 Original commit message from CVS:
12784 * sys/xvimage/xvimagesink.c:
12785 Fix typo that causes the overlay keying color to bright green
12786 on a 16-bit display. Dark grey good. Bright green bad.
12788 2008-04-21 13:47:06 +0000 Stefan Kost <ensonic@users.sourceforge.net>
12790 ext/gnomevfs/gstgnomevfsuri.c: Add FIXME comment about using uri-list for source and sink.
12791 Original commit message from CVS:
12792 * ext/gnomevfs/gstgnomevfsuri.c:
12793 Add FIXME comment about using uri-list for source and sink.
12795 2008-04-20 11:42:37 +0000 Sebastian Dröge <slomo@circular-chaos.org>
12797 ext/ogg/gstogmparse.c: GST_TYPE_FRACTION contains gints so correctly cast gint64 arguments to vaargs functions to gin...
12798 Original commit message from CVS:
12799 * ext/ogg/gstogmparse.c: (gst_ogm_parse_stream_header):
12800 GST_TYPE_FRACTION contains gints so correctly cast gint64 arguments to
12801 vaargs functions to gint. Otherwise the fractions will get 0 set
12802 instead of the correct value on big endian systems. Fixes bug #529018.
12804 2008-04-20 10:17:23 +0000 Sebastian Dröge <slomo@circular-chaos.org>
12806 ext/gnomevfs/: Get the list of supported URI schemes in a threadsafe way and use the same list for the source and sink.
12807 Original commit message from CVS:
12808 * ext/gnomevfs/gstgnomevfssink.c:
12809 (gst_gnome_vfs_sink_uri_get_protocols):
12810 * ext/gnomevfs/gstgnomevfssrc.c:
12811 (gst_gnome_vfs_src_uri_get_protocols):
12812 * ext/gnomevfs/gstgnomevfsuri.c: (_internal_get_supported_uris),
12813 (gst_gnomevfs_get_supported_uris):
12814 Get the list of supported URI schemes in a threadsafe way and use the
12815 same list for the source and sink.
12817 2008-04-20 10:11:54 +0000 Sebastian Dröge <slomo@circular-chaos.org>
12819 ext/gio/gstgio.c: Don't generate a new supported protocols list on each call but cache it. It's supposed to be static...
12820 Original commit message from CVS:
12821 * ext/gio/gstgio.c: (_internal_get_supported_protocols),
12822 (gst_gio_get_supported_protocols):
12823 Don't generate a new supported protocols list on each call but cache
12824 it. It's supposed to be static anyway, this way we only leak it once
12826 * ext/gio/gstgiosink.c: (gst_gio_sink_base_init),
12827 (gst_gio_sink_class_init), (gst_gio_sink_finalize),
12828 (gst_gio_sink_set_property), (gst_gio_sink_get_property),
12829 (gst_gio_sink_start):
12830 * ext/gio/gstgiosink.h:
12831 * ext/gio/gstgiosrc.c: (gst_gio_src_base_init),
12832 (gst_gio_src_class_init), (gst_gio_src_finalize),
12833 (gst_gio_src_set_property), (gst_gio_src_get_property),
12834 (gst_gio_src_start):
12835 * ext/gio/gstgiosrc.h:
12836 API: Add "file" properties where one can set a GFile as source/destination.
12837 Add locking to the properties and use gst_element_class_set_details_simple()
12838 instead of a static GstElementDetails struct.
12840 2008-04-19 20:06:59 +0000 Sebastian Dröge <slomo@circular-chaos.org>
12842 gst/typefind/gsttypefindfunctions.c: Add "mpp" and "mp+" as possible extensions for MusePack files.
12843 Original commit message from CVS:
12844 * gst/typefind/gsttypefindfunctions.c: (musepack_type_find),
12846 Add "mpp" and "mp+" as possible extensions for MusePack files.
12847 Add typefinding for MusePack StreamVersion 8 files and include the
12848 stream version in the caps.
12850 2008-04-19 16:33:24 +0000 Sebastian Dröge <slomo@circular-chaos.org>
12852 gst-libs/gst/rtp/gstrtppayloads.c: Use g_ascii_strcasecmp() instead of the deprecated g_strcasecmp().
12853 Original commit message from CVS:
12854 * gst-libs/gst/rtp/gstrtppayloads.c:
12855 (gst_rtp_payload_info_for_name):
12856 Use g_ascii_strcasecmp() instead of the deprecated g_strcasecmp().
12858 2008-04-18 17:10:43 +0000 Tim-Philipp Müller <tim@centricular.net>
12860 configure.ac: Bump Gtk+ requirement to 2.12.0 for gtk_range_set_fill_level (NB: this only affects compilation of some...
12861 Original commit message from CVS:
12863 Bump Gtk+ requirement to 2.12.0 for gtk_range_set_fill_level
12864 (NB: this only affects compilation of some of the examples).
12865 Remove some configure.ac cruft that's not needed any longer.
12867 2008-04-18 14:54:01 +0000 Edward Hervey <bilboed@bilboed.com>
12869 gst/gdp/gstgdpdepay.c: Don't validate the payload if there isn't any.
12870 Original commit message from CVS:
12871 * gst/gdp/gstgdpdepay.c: (gst_gdp_depay_chain):
12872 Don't validate the payload if there isn't any.
12875 2008-04-17 07:33:46 +0000 Sebastian Dröge <slomo@circular-chaos.org>
12877 gst-libs/gst/audio/gstringbuffer.c: Use g_atomic_int_set() instead of gst_atomic_int_set().
12878 Original commit message from CVS:
12879 * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_may_start):
12880 Use g_atomic_int_set() instead of gst_atomic_int_set().
12882 2008-04-17 07:29:28 +0000 Sebastian Dröge <slomo@circular-chaos.org>
12884 ext/gio/gstgio.c: Return NULL instead of a gchar * array with one NULL element if we don't get any supported URI sche...
12885 Original commit message from CVS:
12886 * ext/gio/gstgio.c: (gst_gio_get_supported_protocols):
12887 Return NULL instead of a gchar * array with one NULL element if we
12888 don't get any supported URI schemes from GIO.
12890 2008-04-15 19:06:00 +0000 Stefan Kost <ensonic@users.sourceforge.net>
12892 gst/audiotestsrc/gstaudiotestsrc.c: Remove cpp style commented old code.
12893 Original commit message from CVS:
12894 * gst/audiotestsrc/gstaudiotestsrc.c:
12895 Remove cpp style commented old code.
12897 2008-04-15 19:02:10 +0000 Stefan Kost <ensonic@users.sourceforge.net>
12899 gst/playback/gstdecodebin2.c: Fix signal docs.
12900 Original commit message from CVS:
12901 * gst/playback/gstdecodebin2.c:
12904 2008-04-14 17:58:19 +0000 Tim-Philipp Müller <tim@centricular.net>
12906 ext/pango/gsttextoverlay.c: Fix textoverlay unit test again by making the supposed default value for the wait-text pr...
12907 Original commit message from CVS:
12908 * ext/pango/gsttextoverlay.c: (gst_text_overlay_class_init),
12909 (gst_text_overlay_init):
12910 Fix textoverlay unit test again by making the supposed default
12911 value for the wait-text property the actual default value.
12912 Also fix Since: tag for new property.
12914 2008-04-11 17:13:52 +0000 Tim-Philipp Müller <tim@centricular.net>
12916 gst-libs/gst/video/video.c: Add guards to these functions to ensure sane input values.
12917 Original commit message from CVS:
12918 * gst-libs/gst/video/video.c: (gst_video_format_new_caps),
12919 (gst_video_format_to_fourcc), (gst_video_format_get_row_stride),
12920 (gst_video_format_get_pixel_stride),
12921 (gst_video_format_get_component_width),
12922 (gst_video_format_get_component_height),
12923 (gst_video_format_get_component_offset), (gst_video_format_get_size),
12924 (gst_video_format_convert):
12925 Add guards to these functions to ensure sane input values.
12926 * tests/check/libs/video.c:
12927 Fix unit test not to create caps with width=0 and height=0.
12929 2008-04-11 01:25:01 +0000 Wim Taymans <wim.taymans@gmail.com>
12931 docs/design/draft-keyframe-force.txt: Fix typo.
12932 Original commit message from CVS:
12933 * docs/design/draft-keyframe-force.txt:
12935 * gst/playback/gstqueue2.c: (update_buffering),
12936 (gst_queue_handle_src_query):
12937 Set buffering mode in the messages.
12938 Set buffering percent in the query.
12939 * tests/examples/seek/seek.c: (update_fill), (msg_state_changed),
12940 (do_stream_buffering), (do_download_buffering), (msg_buffering):
12941 Do some more fancy things based on the buffering method in use.
12943 2008-04-09 21:42:24 +0000 Wim Taymans <wim.taymans@gmail.com>
12945 tests/examples/seek/seek.c: Add basic download reports to seek using the new buffering API.
12946 Original commit message from CVS:
12947 * tests/examples/seek/seek.c: (update_fill), (set_update_fill),
12948 (play_cb), (pause_cb), (stop_cb), (msg_state_changed),
12949 (msg_buffering), (main):
12950 Add basic download reports to seek using the new buffering API.
12952 2008-04-09 21:40:17 +0000 Wim Taymans <wim.taymans@gmail.com>
12954 gst/playback/gstqueue2.c: Include extra buffering stats in the buffering message.
12955 Original commit message from CVS:
12956 * gst/playback/gstqueue2.c: (update_buffering),
12957 (gst_queue_close_temp_location_file), (gst_queue_handle_src_query),
12958 (gst_queue_src_checkgetrange_function):
12959 Include extra buffering stats in the buffering message.
12960 Implement BUFFERING query.
12961 * gst/playback/gsturidecodebin.c: (do_async_start),
12962 (do_async_done), (type_found), (setup_streaming), (setup_source),
12963 (gst_uri_decode_bin_change_state):
12964 Only add decodebin2 when the type is found in streaming mode.
12965 Make uridecodebin async to PAUSED even when we don't have decodebin2
12968 2008-04-09 08:38:19 +0000 Sebastian Dröge <slomo@circular-chaos.org>
12970 ext/gio/gstgio.c: Filter cdda from the supported URI schemes. We can't support musicbrainz tags and everything else o...
12971 Original commit message from CVS:
12972 * ext/gio/gstgio.c: (gst_gio_get_supported_protocols):
12973 Filter cdda from the supported URI schemes. We can't support
12974 musicbrainz tags and everything else one expects from a cdda source
12975 with GIO. Fixes bug #526794.
12977 2008-04-07 22:37:26 +0000 Jan Schmidt <thaytan@mad.scientist.com>
12979 * sys/xvimage/xvimagesink.c:
12980 Fix calculation of 'expected size' for YV12 buffers.
12981 Original commit message from CVS:
12982 2008-04-07 Jan Schmidt <jan.schmidt@sun.com>
12983 * sys/xvimage/xvimagesink.c (gst_xvimagesink_xvimage_new),
12984 (gst_xvimagesink_buffer_alloc):
12985 Fix calculation of 'expected size' for YV12 buffers.
12986 Be a little more verbose in the debug output for buffer-alloc'ed
12987 buffers which turn out to have the wrong size.
12989 2008-04-07 22:26:50 +0000 Jan Schmidt <thaytan@mad.scientist.com>
12992 Fix calculation of 'expected size' for YV12 buffers.
12993 Original commit message from CVS:
12994 * sys/xvimage/xvimagesink.c (gst_xvimagesink_xvimage_new),
12995 (gst_xvimagesink_buffer_alloc):
12996 Fix calculation of 'expected size' for YV12 buffers.
12997 Be a little more verbose in the debug output for buffer-alloc'ed
12998 buffers which turn out to have the wrong size.
13000 2008-04-07 10:50:11 +0000 Tim-Philipp Müller <tim@centricular.net>
13002 Merge other changes from 0.10.19 release branch.
13003 Original commit message from CVS:
13006 * gst-plugins-base.doap:
13007 Merge other changes from 0.10.19 release branch.
13009 2008-04-06 20:16:27 +0000 Tim-Philipp Müller <tim@centricular.net>
13011 gst/: Work around missing bits of thread-safety on older GLibs some more to avoid assertions when starting up multipl...
13012 Original commit message from CVS:
13013 * gst-libs/gst/audio/gstbaseaudiosink.c:
13014 (gst_base_audio_sink_class_init):
13015 * gst-libs/gst/audio/gstbaseaudiosrc.c:
13016 (gst_base_audio_src_class_init):
13017 * gst/playback/gstplayback.c: (plugin_init):
13018 * gst/volume/gstvolume.c: (plugin_init):
13019 Work around missing bits of thread-safety on older GLibs some
13020 more to avoid assertions when starting up multiple playbin
13021 objects concurrently (see #512382).
13023 2008-04-06 17:19:39 +0000 Tim-Philipp Müller <tim@centricular.net>
13025 gst-libs/gst/pbutils/missing-plugins.c: Remove some more fields.
13026 Original commit message from CVS:
13027 * gst-libs/gst/pbutils/missing-plugins.c: (copy_and_clean_caps):
13028 Remove some more fields.
13030 2008-04-06 08:56:07 +0000 Damien Lespiau <damien.lespiau@gmail.com>
13032 configure.ac: Actually build dlls when cross-compiling with mingw32.
13033 Original commit message from CVS:
13034 Patch by: Damien Lespiau <damien dot lespiau at gmail dot com>
13036 Actually build dlls when cross-compiling with mingw32.
13039 2008-04-03 23:01:11 +0000 Tim-Philipp Müller <tim@centricular.net>
13041 configure.ac: Bump version to 0.10.19.1 after the unplanned 0.10.19 release.
13042 Original commit message from CVS:
13044 Bump version to 0.10.19.1 after the unplanned 0.10.19 release.
13046 2008-04-03 16:10:53 +0000 Wim Taymans <wim.taymans@gmail.com>
13048 tests/examples/seek/seek.c: Add statusbar.
13049 Original commit message from CVS:
13050 * tests/examples/seek/seek.c: (play_cb), (pause_cb), (stop_cb),
13051 (msg_buffering), (connect_bus_signals), (main):
13053 Add buffering support with feedback in the statusbar.
13055 2008-04-03 15:58:37 +0000 Tim-Philipp Müller <tim@centricular.net>
13057 ext/ogg/gstoggmux.c: Fix sample pipeline description.
13058 Original commit message from CVS:
13059 * ext/ogg/gstoggmux.c:
13060 Fix sample pipeline description.
13062 2008-04-03 14:58:06 +0000 Stefan Kost <ensonic@users.sourceforge.net>
13064 docs/plugins/: Add playbin, playbin2, decodebin, decodebin2, uridecodebin and oggmux
13065 Original commit message from CVS:
13066 * docs/plugins/Makefile.am:
13067 * docs/plugins/gst-plugins-base-plugins-docs.sgml:
13068 * docs/plugins/gst-plugins-base-plugins-overrides.txt:
13069 * docs/plugins/gst-plugins-base-plugins-sections.txt:
13070 Add playbin, playbin2, decodebin, decodebin2, uridecodebin and oggmux
13071 * docs/plugins/gst-plugins-base-plugins.args:
13072 * docs/plugins/gst-plugins-base-plugins.hierarchy:
13073 * docs/plugins/gst-plugins-base-plugins.interfaces:
13074 * docs/plugins/gst-plugins-base-plugins.prerequisites:
13075 * docs/plugins/inspect/plugin-adder.xml:
13076 * docs/plugins/inspect/plugin-alsa.xml:
13077 * docs/plugins/inspect/plugin-audioconvert.xml:
13078 * docs/plugins/inspect/plugin-audiorate.xml:
13079 * docs/plugins/inspect/plugin-audioresample.xml:
13080 * docs/plugins/inspect/plugin-audiotestsrc.xml:
13081 * docs/plugins/inspect/plugin-cdparanoia.xml:
13082 * docs/plugins/inspect/plugin-decodebin.xml:
13083 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
13084 * docs/plugins/inspect/plugin-gdp.xml:
13085 * docs/plugins/inspect/plugin-gnomevfs.xml:
13086 * docs/plugins/inspect/plugin-libvisual.xml:
13087 * docs/plugins/inspect/plugin-ogg.xml:
13088 * docs/plugins/inspect/plugin-pango.xml:
13089 * docs/plugins/inspect/plugin-playback.xml:
13090 * docs/plugins/inspect/plugin-queue2.xml:
13091 * docs/plugins/inspect/plugin-subparse.xml:
13092 * docs/plugins/inspect/plugin-tcp.xml:
13093 * docs/plugins/inspect/plugin-theora.xml:
13094 * docs/plugins/inspect/plugin-typefindfunctions.xml:
13095 * docs/plugins/inspect/plugin-uridecodebin.xml:
13096 * docs/plugins/inspect/plugin-video4linux.xml:
13097 * docs/plugins/inspect/plugin-videorate.xml:
13098 * docs/plugins/inspect/plugin-videoscale.xml:
13099 * docs/plugins/inspect/plugin-videotestsrc.xml:
13100 * docs/plugins/inspect/plugin-volume.xml:
13101 * docs/plugins/inspect/plugin-vorbis.xml:
13102 * docs/plugins/inspect/plugin-ximagesink.xml:
13103 * docs/plugins/inspect/plugin-xvimagesink.xml:
13104 Update introspection data.
13105 * ext/ogg/gstoggmux.c:
13107 * gst/playback/gstdecodebin2.c:
13108 Don't use gtk-doc style comment start for private stuff, but make it
13109 formatted like this for consistency.
13111 2008-04-03 12:16:04 +0000 Wim Taymans <wim.taymans@gmail.com>
13113 gst/playback/gstdecodebin2.c: Remove fakesink hack, we can now implement this more elegantly.
13114 Original commit message from CVS:
13115 * gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init),
13116 (gst_decode_bin_init), (gst_decode_bin_dispose),
13117 (gst_decode_bin_set_sink_caps), (gst_decode_bin_get_sink_caps),
13118 (gst_decode_bin_set_property), (gst_decode_bin_get_property),
13119 (analyze_new_pad), (connect_pad), (expose_pad),
13120 (gst_decode_group_new), (gst_decode_group_control_demuxer_pad),
13121 (gst_decode_group_expose), (gst_decode_group_free),
13122 (do_async_start), (do_async_done), (gst_decode_bin_change_state):
13123 Remove fakesink hack, we can now implement this more elegantly.
13124 Added property to bypass typefinding.
13125 Removed underrun callback and demuxer pad probe, we now use the srcpad
13126 probe to expose groups.
13127 API::sink-caps property
13128 * gst/playback/gstplaybin2.c: (no_more_pads_cb):
13129 Guard against multiple emissions of the no_more_pads signal, which
13130 happens when we are dealing with chained oggs.
13131 * gst/playback/gsturidecodebin.c: (remove_decoders),
13132 (make_decoder), (type_found), (setup_streaming), (source_new_pad),
13134 For streams, use our own typefind element and plug our queue after it.
13135 We will need this to determine the type of buffering to use for the
13138 2008-04-03 10:37:03 +0000 Wim Taymans <wim.taymans@gmail.com>
13140 gst-libs/gst/audio/gstbaseaudiosink.c: Guard against over and underflows because of clock slaving.
13141 Original commit message from CVS:
13142 * gst-libs/gst/audio/gstbaseaudiosink.c:
13143 (gst_base_audio_sink_skew_slaving), (gst_base_audio_sink_render):
13144 Guard against over and underflows because of clock slaving.
13145 When we are using our own clock, still compensate for any calibrations
13146 that we might have done to our clock.
13148 2008-04-03 10:22:33 +0000 Wim Taymans <wim.taymans@gmail.com>
13150 ext/theora/theoradec.c: Don't try to do anything fancy with the return code from pushing an event, it does not have e...
13151 Original commit message from CVS:
13152 * ext/theora/theoradec.c: (theora_handle_type_packet),
13153 (theora_dec_chain):
13154 Don't try to do anything fancy with the return code from pushing an
13155 event, it does not have enough information to turn it into a
13158 2008-04-03 10:19:43 +0000 Wim Taymans <wim.taymans@gmail.com>
13160 ext/ogg/gstoggdemux.c: Add small debug line.
13161 Original commit message from CVS:
13162 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_reset),
13163 (gst_ogg_demux_chain_elem_pad):
13164 Add small debug line.
13165 Pass return code from the internal decoder instead of the too generic
13168 2008-04-03 06:39:27 +0000 Sebastian Dröge <slomo@circular-chaos.org>
13170 gst-libs/gst/cdda/: Use GLib's base64 implementation instead of our own.
13171 Original commit message from CVS:
13172 * gst-libs/gst/cdda/Makefile.am:
13173 * gst-libs/gst/cdda/base64.c:
13174 * gst-libs/gst/cdda/base64.h:
13175 * gst-libs/gst/cdda/gstcddabasesrc.c:
13176 (gst_cddabasesrc_calculate_musicbrainz_discid):
13177 Use GLib's base64 implementation instead of our own.
13179 2008-04-02 15:41:50 +0000 Wim Taymans <wim.taymans@gmail.com>
13181 ext/ogg/gstoggdemux.c: Refix oggdemux, we only have a problem if we failed to find a chain and we are not EOF.
13182 Original commit message from CVS:
13183 * ext/ogg/gstoggdemux.c: (gst_ogg_demux_activate_chain),
13184 (gst_ogg_demux_read_chain):
13185 Refix oggdemux, we only have a problem if we failed to find a chain and
13188 2008-04-02 15:07:01 +0000 Victor STINNER <victor.stinner@haypocalc.com>
13190 ext/ogg/gstoggdemux.c: When we fail to find a BOS page and we and up with no chain, error out properly instead of seg...
13191 Original commit message from CVS:
13192 Patch by: Victor STINNER <victor dot stinner at haypocalc dot com>
13193 * ext/ogg/gstoggdemux.c: (gst_ogg_demux_activate_chain),
13194 (gst_ogg_demux_read_chain):
13195 When we fail to find a BOS page and we and up with no chain, error out
13196 properly instead of segfaulting. Fixes #525665.
13198 2008-04-02 14:58:05 +0000 Wim Taymans <wim.taymans@gmail.com>
13200 ext/ogg/gstoggdemux.c: The new-pad-group sequence is add-pads, no-more-pads, add-pads, no-more-pads...
13201 Original commit message from CVS:
13202 * ext/ogg/gstoggdemux.c: (gst_ogg_demux_activate_chain),
13203 (gst_ogg_demux_read_chain), (gst_ogg_demux_handle_page):
13204 The new-pad-group sequence is add-pads, no-more-pads, add-pads,
13207 2008-04-02 11:08:05 +0000 Wim Taymans <wim.taymans@gmail.com>
13209 gst/playback/gstqueue2.c: Update the estimated input data when we push out a buffer.
13210 Original commit message from CVS:
13211 * gst/playback/gstqueue2.c: (update_out_rates),
13212 (gst_queue_open_temp_location_file),
13213 (gst_queue_close_temp_location_file), (gst_queue_handle_src_event),
13214 (gst_queue_handle_src_query), (gst_queue_set_property):
13215 Update the estimated input data when we push out a buffer.
13216 Add some debug info about the temp file.
13217 Only forward src events when we are not using a temp file.
13218 Don't block the duration query, we need to find something better.
13219 Don't leak the temp filename.
13221 2008-04-01 14:01:14 +0000 Sebastian Dröge <slomo@circular-chaos.org>
13223 configure.ac: Require GLib 2.12 and liboil 0.3.14.
13224 Original commit message from CVS:
13226 Require GLib 2.12 and liboil 0.3.14.
13227 * gst/volume/gstvolume.c: (volume_process_double):
13228 Unconditionally use liboil 0.3.14 function.
13230 2008-03-31 16:08:45 +0000 Wim Taymans <wim.taymans@gmail.com>
13232 gst-libs/gst/riff/riff-media.c: ms-gsm can have arbitrarty sample rates. See #481354.
13233 Original commit message from CVS:
13234 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
13235 ms-gsm can have arbitrarty sample rates. See #481354.
13237 2008-03-28 16:22:35 +0000 Wim Taymans <wim.taymans@gmail.com>
13239 gst-libs/gst/riff/riff-media.c: MP4S is generic MPEG-4, not a microsoft variant.
13240 Original commit message from CVS:
13241 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
13242 MP4S is generic MPEG-4, not a microsoft variant.
13244 2008-03-27 15:26:38 +0000 Michael Smith <msmith@xiph.org>
13246 gst/gdp/gstgdpdepay.c: Check the body CRC (if set) when depayloading.
13247 Original commit message from CVS:
13248 * gst/gdp/gstgdpdepay.c: (gst_gdp_depay_chain):
13249 Check the body CRC (if set) when depayloading.
13252 2008-03-24 17:45:36 +0000 Tim-Philipp Müller <tim@centricular.net>
13254 ext/pango/gsttextoverlay.c: Fix Since: version for new property.
13255 Original commit message from CVS:
13256 * ext/pango/gsttextoverlay.c: (gst_text_overlay_class_init):
13257 Fix Since: version for new property.
13259 2008-03-24 16:40:08 +0000 Wim Taymans <wim.taymans@gmail.com>
13261 gst-libs/gst/rtsp/gstrtspconnection.c: Don't error when poll_wait returns EAGAIN.
13262 Original commit message from CVS:
13263 * gst-libs/gst/rtsp/gstrtspconnection.c:
13264 (gst_rtsp_connection_connect), (gst_rtsp_connection_write),
13265 (gst_rtsp_connection_read_internal), (gst_rtsp_connection_poll):
13266 Don't error when poll_wait returns EAGAIN.
13268 2008-03-24 14:08:22 +0000 Wim Taymans <wim.taymans@gmail.com>
13270 gst/playback/gstqueue2.c: The queue is never filled when there are no buffers in the queue at all.
13271 Original commit message from CVS:
13272 * gst/playback/gstqueue2.c: (gst_queue_is_filled):
13273 The queue is never filled when there are no buffers in the queue at all.
13276 2008-03-24 12:26:30 +0000 Wim Taymans <wim.taymans@gmail.com>
13278 gst/playback/gstplaybin2.c: Update some docs.
13279 Original commit message from CVS:
13280 * gst/playback/gstplaybin2.c: (gst_play_bin_class_init),
13281 (init_group), (free_group), (gst_play_bin_init),
13282 (gst_play_bin_finalize), (gst_play_bin_set_uri),
13283 (gst_play_bin_set_suburi), (gst_play_bin_get_video_tags),
13284 (gst_play_bin_get_audio_tags), (gst_play_bin_get_text_tags),
13285 (gst_play_bin_set_current_video_stream),
13286 (gst_play_bin_set_current_audio_stream),
13287 (gst_play_bin_set_current_text_stream),
13288 (gst_play_bin_set_encoding), (gst_play_bin_set_property),
13289 (gst_play_bin_get_property), (pad_added_cb), (pad_removed_cb),
13290 (no_more_pads_cb), (perform_eos), (autoplug_select_cb),
13291 (activate_group), (deactivate_group), (setup_next_source),
13292 (save_current_group), (gst_play_bin_change_state):
13294 Add new locks and conds to protect pipeline creation and group
13296 Implement the sub-uri property.
13297 Keep track of pending uridecodebin creation and configure the output
13298 pipeline after all streams are configured.
13299 Propagate subtitle encoding to the uridecodebins.
13300 Implement getting the video/audio/visualisation elements.
13301 Use input-selector for stream switching.
13302 If we are asked to do visualisation, prefer to autoplug raw sinks
13303 instead of sinks that accept encoded data.
13305 2008-03-24 12:15:26 +0000 Wim Taymans <wim.taymans@gmail.com>
13307 gst/playback/gstplaysink.*: Add methods to get audio/video/vis elements.
13308 Original commit message from CVS:
13309 * gst/playback/gstplaysink.c: (gst_play_sink_class_init),
13310 (gst_play_sink_init), (gst_play_sink_dispose),
13311 (gst_play_sink_set_video_sink), (gst_play_sink_get_video_sink),
13312 (gst_play_sink_set_audio_sink), (gst_play_sink_get_audio_sink),
13313 (gst_play_sink_vis_unblocked), (gst_play_sink_vis_blocked),
13314 (gst_play_sink_set_vis_plugin), (gst_play_sink_get_vis_plugin),
13315 (gst_play_sink_set_volume), (gst_play_sink_get_volume),
13316 (gst_play_sink_set_mute), (gen_video_chain), (gen_text_chain),
13317 (gen_audio_chain), (gen_vis_chain), (gst_play_sink_reconfigure),
13318 (gst_play_sink_set_font_desc), (gst_play_sink_get_font_desc),
13319 (gst_play_sink_send_event_to_sink), (gst_play_sink_change_state):
13320 * gst/playback/gstplaysink.h:
13321 Add methods to get audio/video/vis elements.
13322 Add methods to set the font description for the overlay.
13323 Remove properties, we're using this element with its methods only.
13324 Add support for subtitles.
13325 Rearrange the locking a bit to not use the object lock for protecting
13326 the pipeline construction.
13327 Try to use the volume and mute property on the sink when its available.
13328 Implement the mute option with volume when the sink does not have a mute
13330 Only add volume element when the sink has no volume property.
13331 Only do visualisations with raw audio pads.
13333 2008-03-24 12:03:02 +0000 Wim Taymans <wim.taymans@gmail.com>
13335 ext/pango/gsttextoverlay.*: Add property to configure waiting for text on the textpad or not, with the default behavi...
13336 Original commit message from CVS:
13337 * ext/pango/gsttextoverlay.c: (gst_text_overlay_class_init),
13338 (gst_text_overlay_init), (gst_text_overlay_set_property),
13339 (gst_text_overlay_get_property), (gst_text_overlay_src_event),
13340 (gst_text_overlay_text_event), (gst_text_overlay_video_event),
13341 (gst_text_overlay_text_chain), (gst_text_overlay_video_chain),
13342 (gst_text_overlay_change_state):
13343 * ext/pango/gsttextoverlay.h:
13344 Add property to configure waiting for text on the textpad or not, with
13345 the default behaviour being the old one (always wait for text before
13346 rendering the video). This default behaviour is usually not the best one
13347 because the text stream can very sparse and could require queueing a lot
13349 Fix the flushing and EOS handing so that we don't mix up their meaning.
13351 2008-03-24 11:54:02 +0000 Wim Taymans <wim.taymans@gmail.com>
13353 gst/playback/gsturidecodebin.c: Add a readonly source property and notify.
13354 Original commit message from CVS:
13355 * gst/playback/gsturidecodebin.c:
13356 (gst_uri_decode_bin_autoplug_factories),
13357 (gst_uri_decode_bin_class_init), (gst_uri_decode_bin_init),
13358 (gst_uri_decode_bin_finalize), (gst_uri_decode_bin_set_encoding),
13359 (gst_uri_decode_bin_set_property),
13360 (gst_uri_decode_bin_get_property), (no_more_pads_full),
13361 (new_decoded_pad_cb), (gen_source_element), (remove_decoders),
13362 (proxy_autoplug_factories_signal), (make_decoder),
13363 (source_new_pad), (setup_source):
13364 Add a readonly source property and notify.
13365 Add new lock for protecting the construction of the pipeline.
13366 Keep track of the decodebins we plugged.
13367 Correctly proxy the autoplug signal so that it actually continues.
13368 Proxy subtitle-encoding to the decodebins.
13370 2008-03-24 11:46:15 +0000 Wim Taymans <wim.taymans@gmail.com>
13372 tests/examples/seek/seek.c: Rearrange some buttons in playbin2 and make some other boxes insensitive when needed.
13373 Original commit message from CVS:
13374 * tests/examples/seek/seek.c: (audio_toggle_cb), (video_toggle_cb),
13375 (text_toggle_cb), (update_streams), (main):
13376 Rearrange some buttons in playbin2 and make some other boxes insensitive
13378 Add language codes to subtitle selection boxes when we gind the right
13379 tags for the streams.
13381 2008-03-24 11:36:08 +0000 Wim Taymans <wim.taymans@gmail.com>
13383 gst/playback/gstdecodebin2.c: Protect caps property with the object lock.
13384 Original commit message from CVS:
13385 * gst/playback/gstdecodebin2.c: (gst_decode_bin_dispose),
13386 (gst_decode_bin_set_caps), (gst_decode_bin_get_caps),
13387 (gst_decode_bin_set_subs_encoding),
13388 (gst_decode_bin_get_subs_encoding),
13389 (gst_decode_bin_autoplug_factories), (connect_pad), (are_raw_caps),
13390 (deactivate_free_recursive):
13391 Protect caps property with the object lock.
13392 Protect encoding property with the object lock.
13393 Keep list of elements we added that have the subtitle-encoding property.
13394 Distribute the subtitle-encoding to all of the elements when it
13397 2008-03-24 11:24:22 +0000 Wim Taymans <wim.taymans@gmail.com>
13399 gst-libs/gst/audio/gstaudiosink.c: Small debug improvement.
13400 Original commit message from CVS:
13401 * gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_release):
13402 Small debug improvement.
13403 * gst-libs/gst/audio/gstbaseaudiosink.c:
13404 (gst_base_audio_sink_render):
13405 Fix bug in determining the sample start/stop position, we want to base
13406 this decision on the fact that we are going forwards or backwards, not
13407 slower or faster. This fixes some ugly resync warnings when playing at
13410 2008-03-23 13:41:28 +0000 Sebastian Dröge <slomo@circular-chaos.org>
13412 ext/gio/gstgio.c: Correctly set the supported URI schemes and don't leave some schemes in the middle or at the start ...
13413 Original commit message from CVS:
13414 * ext/gio/gstgio.c: (gst_gio_get_supported_protocols):
13415 Correctly set the supported URI schemes and don't leave
13416 some schemes in the middle or at the start at NULL.
13418 2008-03-23 13:12:41 +0000 Tim-Philipp Müller <tim@centricular.net>
13420 tests/check/elements/gdpdepay.c: Make test compile without unused function/variable warnings on PPC.
13421 Original commit message from CVS:
13422 * tests/check/elements/gdpdepay.c:
13423 Make test compile without unused function/variable warnings on PPC.
13425 2008-03-22 15:00:53 +0000 Sebastian Dröge <slomo@circular-chaos.org>
13427 Use G_PARAM_STATIC_STRINGS everywhere for GParamSpecs that use static strings (i.e. all). This gives us less memory u...
13428 Original commit message from CVS:
13430 * ext/alsa/gstalsamixerelement.c:
13431 (gst_alsa_mixer_element_class_init):
13432 * ext/alsa/gstalsasink.c: (gst_alsasink_class_init):
13433 * ext/alsa/gstalsasrc.c: (gst_alsasrc_class_init):
13434 * ext/cdparanoia/gstcdparanoiasrc.c:
13435 (gst_cd_paranoia_src_class_init):
13436 * ext/gio/gstgiosink.c: (gst_gio_sink_class_init):
13437 * ext/gio/gstgiosrc.c: (gst_gio_src_class_init):
13438 * ext/gio/gstgiostreamsink.c: (gst_gio_stream_sink_class_init):
13439 * ext/gio/gstgiostreamsrc.c: (gst_gio_stream_src_class_init):
13440 * ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_class_init):
13441 * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_class_init):
13442 * ext/ogg/gstoggmux.c: (gst_ogg_mux_class_init):
13443 * ext/pango/gsttextoverlay.c: (gst_text_overlay_class_init):
13444 * ext/pango/gsttextrender.c: (gst_text_render_class_init):
13445 * ext/theora/theoradec.c: (gst_theora_dec_class_init):
13446 * ext/theora/theoraenc.c: (gst_theora_enc_class_init):
13447 * ext/theora/theoraparse.c: (gst_theora_parse_class_init):
13448 * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_class_init):
13449 * gst-libs/gst/audio/gstaudiofiltertemplate.c:
13450 (gst_audio_filter_template_class_init):
13451 * gst-libs/gst/audio/gstbaseaudiosink.c:
13452 (gst_base_audio_sink_class_init):
13453 * gst-libs/gst/audio/gstbaseaudiosrc.c:
13454 (gst_base_audio_src_class_init):
13455 * gst-libs/gst/cdda/gstcddabasesrc.c:
13456 (gst_cdda_base_src_class_init):
13457 * gst-libs/gst/interfaces/mixertrack.c:
13458 (gst_mixer_track_class_init):
13459 * gst-libs/gst/rtp/gstbasertpdepayload.c:
13460 (gst_base_rtp_depayload_class_init):
13461 * gst-libs/gst/rtp/gstbasertppayload.c:
13462 (gst_basertppayload_class_init):
13463 * gst/audioconvert/gstaudioconvert.c:
13464 (gst_audio_convert_class_init):
13465 * gst/audiorate/gstaudiorate.c: (gst_audio_rate_class_init):
13466 * gst/audioresample/gstaudioresample.c:
13467 (gst_audioresample_class_init):
13468 * gst/audiotestsrc/gstaudiotestsrc.c:
13469 (gst_audio_test_src_class_init):
13470 * gst/gdp/gstgdppay.c: (gst_gdp_pay_class_init):
13471 * gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init):
13472 * gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init),
13473 (preroll_unlinked):
13474 * gst/playback/gstplaybin.c: (gst_play_bin_class_init):
13475 * gst/playback/gstplaybin2.c: (gst_play_bin_class_init):
13476 * gst/playback/gstplaysink.c: (gst_play_sink_class_init):
13477 * gst/playback/gstqueue2.c: (gst_queue_class_init):
13478 * gst/playback/gststreaminfo.c: (gst_stream_info_class_init):
13479 * gst/playback/gststreamselector.c: (gst_selector_pad_class_init),
13480 (gst_stream_selector_class_init):
13481 * gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_class_init):
13482 * gst/subparse/gstsubparse.c: (gst_sub_parse_class_init):
13483 * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init):
13484 * gst/tcp/gsttcpclientsink.c: (gst_tcp_client_sink_class_init):
13485 * gst/tcp/gsttcpclientsrc.c: (gst_tcp_client_src_class_init):
13486 * gst/tcp/gsttcpserversink.c: (gst_tcp_server_sink_class_init):
13487 * gst/tcp/gsttcpserversrc.c: (gst_tcp_server_src_class_init):
13488 * gst/videorate/gstvideorate.c: (gst_video_rate_class_init):
13489 * gst/videoscale/gstvideoscale.c: (gst_video_scale_class_init):
13490 * gst/videotestsrc/gstvideotestsrc.c:
13491 (gst_video_test_src_class_init):
13492 * gst/volume/gstvolume.c: (gst_volume_class_init):
13493 * sys/v4l/gstv4lelement.c: (gst_v4lelement_class_init):
13494 * sys/v4l/gstv4lmjpegsink.c: (gst_v4lmjpegsink_class_init):
13495 * sys/v4l/gstv4lmjpegsrc.c: (gst_v4lmjpegsrc_class_init):
13496 * sys/v4l/gstv4lsrc.c: (gst_v4lsrc_class_init):
13497 * sys/ximage/ximagesink.c: (gst_ximagesink_class_init):
13498 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_class_init):
13499 Use G_PARAM_STATIC_STRINGS everywhere for GParamSpecs that use
13500 static strings (i.e. all). This gives us less memory usage,
13501 fewer allocations and thus less memory defragmentation. Depend
13502 on core CVS for this. Fixes bug #523806.
13504 2008-03-22 14:13:55 +0000 Sebastian Dröge <slomo@circular-chaos.org>
13506 ext/gio/gstgio.c: Filter http and https protocols. GIO/GVfs handles them but it's impossible to implement iradio/icec...
13507 Original commit message from CVS:
13508 * ext/gio/gstgio.c: (gst_gio_get_supported_protocols):
13509 Filter http and https protocols. GIO/GVfs handles them but it's
13510 impossible to implement iradio/icecast with it. Better use
13511 souphttpsrc or something else for this.
13512 * ext/gio/gstgiobasesrc.c: (gst_gio_base_src_get_size):
13513 If getting the file informations by a query fails try it with the
13514 seek-to-end trick too.
13516 2008-03-21 16:46:33 +0000 Sebastian Dröge <slomo@circular-chaos.org>
13518 gst/volume/gstvolume.c: memset buffers to zero if we get a GAP buffer. We usually see a buffer as one unit so let's h...
13519 Original commit message from CVS:
13520 * gst/volume/gstvolume.c: (gst_volume_interface_supported),
13521 (gst_volume_base_init), (gst_volume_class_init),
13522 (volume_process_double), (volume_process_float),
13523 (volume_transform_ip), (plugin_init):
13524 memset buffers to zero if we get a GAP buffer. We usually see a
13525 buffer as one unit so let's handle it as one and don't care about
13526 volume changes while processing one buffer.
13527 Also clean up some stuff a bit.
13529 2008-03-21 15:58:44 +0000 Sebastian Dröge <slomo@circular-chaos.org>
13531 gst/audioconvert/gstaudioconvert.c: Make audioconvert GAP-aware by outputting silence buffers when the input has the ...
13532 Original commit message from CVS:
13533 * gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_init),
13534 (gst_audio_convert_create_silence_buffer),
13535 (gst_audio_convert_transform):
13536 Make audioconvert GAP-aware by outputting silence buffers when the
13537 input has the GAP flag set. This is up to 8x faster.
13538 Based on a patch by Stefan Kost. Fixes bug #517813.
13540 2008-03-21 15:54:54 +0000 Sebastian Dröge <slomo@circular-chaos.org>
13542 gst/volume/gstvolume.c: Use oil_scalarmultiply_f64_ns() for double processing when it's available at compile time.
13543 Original commit message from CVS:
13544 * gst/volume/gstvolume.c: (volume_process_double):
13545 Use oil_scalarmultiply_f64_ns() for double processing when it's
13546 available at compile time.
13548 2008-03-21 13:27:47 +0000 Sebastian Dröge <slomo@circular-chaos.org>
13550 configure.ac: Fix lrint/lrintf checks to actually work. These functions are in libm on Linux at least so try to link ...
13551 Original commit message from CVS:
13553 Fix lrint/lrintf checks to actually work. These functions are
13554 in libm on Linux at least so try to link to it.
13556 2008-03-21 00:36:20 +0000 Jan Schmidt <thaytan@mad.scientist.com>
13558 configure.ac: Back to development - 0.10.18.1
13559 Original commit message from CVS:
13561 Back to development - 0.10.18.1
13563 === release 0.10.18 ===
13565 2008-03-21 00:26:03 +0000 Jan Schmidt <thaytan@mad.scientist.com>
13571 * docs/plugins/gst-plugins-base-plugins.args:
13572 * docs/plugins/gst-plugins-base-plugins.hierarchy:
13573 * docs/plugins/gst-plugins-base-plugins.interfaces:
13574 * docs/plugins/gst-plugins-base-plugins.prerequisites:
13575 * docs/plugins/gst-plugins-base-plugins.signals:
13576 * docs/plugins/inspect/plugin-adder.xml:
13577 * docs/plugins/inspect/plugin-alsa.xml:
13578 * docs/plugins/inspect/plugin-audioconvert.xml:
13579 * docs/plugins/inspect/plugin-audiorate.xml:
13580 * docs/plugins/inspect/plugin-audioresample.xml:
13581 * docs/plugins/inspect/plugin-audiotestsrc.xml:
13582 * docs/plugins/inspect/plugin-cdparanoia.xml:
13583 * docs/plugins/inspect/plugin-decodebin.xml:
13584 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
13585 * docs/plugins/inspect/plugin-gdp.xml:
13586 * docs/plugins/inspect/plugin-gnomevfs.xml:
13587 * docs/plugins/inspect/plugin-libvisual.xml:
13588 * docs/plugins/inspect/plugin-ogg.xml:
13589 * docs/plugins/inspect/plugin-pango.xml:
13590 * docs/plugins/inspect/plugin-playback.xml:
13591 * docs/plugins/inspect/plugin-queue2.xml:
13592 * docs/plugins/inspect/plugin-subparse.xml:
13593 * docs/plugins/inspect/plugin-tcp.xml:
13594 * docs/plugins/inspect/plugin-theora.xml:
13595 * docs/plugins/inspect/plugin-typefindfunctions.xml:
13596 * docs/plugins/inspect/plugin-uridecodebin.xml:
13597 * docs/plugins/inspect/plugin-video4linux.xml:
13598 * docs/plugins/inspect/plugin-videorate.xml:
13599 * docs/plugins/inspect/plugin-videoscale.xml:
13600 * docs/plugins/inspect/plugin-videotestsrc.xml:
13601 * docs/plugins/inspect/plugin-volume.xml:
13602 * docs/plugins/inspect/plugin-vorbis.xml:
13603 * docs/plugins/inspect/plugin-ximagesink.xml:
13604 * docs/plugins/inspect/plugin-xvimagesink.xml:
13605 * gst-plugins-base.doap:
13607 * win32/common/config.h:
13609 Original commit message from CVS:
13612 2008-03-21 00:16:37 +0000 Jan Schmidt <thaytan@mad.scientist.com>
13639 Original commit message from CVS:
13642 2008-03-18 12:19:43 +0000 Jan Schmidt <thaytan@mad.scientist.com>
13644 0.10.17.4 pre-release
13645 Original commit message from CVS:
13647 * win32/common/config.h:
13648 0.10.17.4 pre-release
13650 2008-03-18 11:20:05 +0000 Wim Taymans <wim.taymans@gmail.com>
13652 gst-libs/gst/sdp/gstsdpmessage.c: Use GST_STR_NULL when trying to print strings that could be NULL because this might...
13653 Original commit message from CVS:
13654 * gst-libs/gst/sdp/gstsdpmessage.c: (gst_sdp_message_dump):
13655 Use GST_STR_NULL when trying to print strings that could be NULL because
13656 this might crash on some platforms. See #520808.
13658 2008-03-18 11:10:12 +0000 Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>
13660 gst-libs/gst/rtsp/gstrtspconnection.c: Generic Windows fixes that makes libgstrtsp work on Windows when coupled with ...
13661 Original commit message from CVS:
13662 Patch by: Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>
13663 * gst-libs/gst/rtsp/gstrtspconnection.c:
13664 (gst_rtsp_connection_connect), (gst_rtsp_connection_write),
13665 (read_line), (gst_rtsp_connection_read_internal):
13666 Generic Windows fixes that makes libgstrtsp work on Windows when
13667 coupled with the new GstPoll API. See #520808.
13669 2008-03-17 22:06:56 +0000 Milosz Derezynski <internalerror@gmail.com>
13671 ext/gio/gstgiobasesrc.c: If seeking to a new position succeeds don't simply return from create() without creating a b...
13672 Original commit message from CVS:
13673 Patch by: Milosz Derezynski <internalerror at gmail dot com>
13674 * ext/gio/gstgiobasesrc.c: (gst_gio_base_src_create):
13675 If seeking to a new position succeeds don't simply return from
13676 create() without creating a buffer. Do this only in the case
13677 seeking to the new position fails. Fixes bug #523054.
13679 2008-03-17 10:32:28 +0000 Tim-Philipp Müller <tim@centricular.net>
13681 gst-libs/gst/video/video.c: Fix gst_video_format_parse_caps() for RGB caps with alpha channel (#522635).
13682 Original commit message from CVS:
13683 * gst-libs/gst/video/video.c: (gst_video_format_parse_caps),
13684 (gst_video_format_from_rgba32_masks):
13685 Fix gst_video_format_parse_caps() for RGB caps with alpha channel
13687 * tests/check/libs/video.c: (test_parse_caps_rgb), (video_suite):
13688 Add unit test for the RGB caps parsing and creation, checking for
13689 internal consistency of the new API and consistency of the API with
13690 the old GST_VIDEO_CAPS_* defines.
13692 2008-03-14 18:42:35 +0000 David Schleef <ds@schleef.org>
13694 gst/videotestsrc/videotestsrc.c: Oops, revert last change because -base is in freeze.
13695 Original commit message from CVS:
13696 * gst/videotestsrc/videotestsrc.c: Oops, revert last change
13697 because -base is in freeze.
13699 2008-03-14 17:33:09 +0000 William M. Brack <wbrack@mmm.hk>
13701 gst/videotestsrc/videotestsrc.c: Fix Bayer pattern generation.
13702 Original commit message from CVS:
13703 Patch by: William M. Brack
13704 * gst/videotestsrc/videotestsrc.c: Fix Bayer pattern generation.
13706 2008-03-14 09:54:44 +0000 Wim Taymans <wim.taymans@gmail.com>
13708 gst/playback/gststreamselector.*: Revert change that caused regression until a real fix is found.
13709 Original commit message from CVS:
13710 * gst/playback/gststreamselector.c: (gst_selector_pad_event),
13711 (gst_selector_pad_chain):
13712 * gst/playback/gststreamselector.h:
13713 Revert change that caused regression until a real fix is found.
13716 2008-03-12 12:39:13 +0000 Michael Smith <msmith@xiph.org>
13718 gst-libs/gst/audio/gstringbuffer.*: Rename recently added buffer types to make more sense.
13719 Original commit message from CVS:
13720 * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_parse_caps):
13721 * gst-libs/gst/audio/gstringbuffer.h:
13722 Rename recently added buffer types to make more sense.
13723 * ext/alsa/gstalsasink.c: (alsasink_parse_spec),
13724 (gst_alsasink_write):
13725 Adapt for above API changes.
13728 2008-03-11 13:23:55 +0000 Sebastian Dröge <slomo@circular-chaos.org>
13730 win32/common/libgstnetbuffer.def: Add new symbol gst_netaddress_equal. Fixes bug #521743.
13731 Original commit message from CVS:
13732 * win32/common/libgstnetbuffer.def:
13733 Add new symbol gst_netaddress_equal. Fixes bug #521743.
13735 2008-03-11 00:25:13 +0000 Jan Schmidt <thaytan@mad.scientist.com>
13737 0.10.17.3 pre-release
13738 Original commit message from CVS:
13740 * win32/common/config.h:
13741 0.10.17.3 pre-release
13743 2008-03-10 17:19:56 +0000 Wim Taymans <wim.taymans@gmail.com>
13745 gst-libs/gst/audio/gstbaseaudiosrc.c: Fix duration when no clock was provided. Fixes #520300.
13746 Original commit message from CVS:
13747 * gst-libs/gst/audio/gstbaseaudiosrc.c:
13748 (gst_base_audio_src_create):
13749 Fix duration when no clock was provided. Fixes #520300.
13751 2008-03-07 18:17:44 +0000 Olivier Crete <tester@tester.ca>
13753 Add trivial function to compare GstNetAddress. See #520626.
13754 Original commit message from CVS:
13755 Patch by: Olivier Crete <tester at tester ca>
13756 * docs/libs/gst-plugins-base-libs-sections.txt:
13757 * gst-libs/gst/netbuffer/gstnetbuffer.c: (gst_netaddress_equal):
13758 * gst-libs/gst/netbuffer/gstnetbuffer.h:
13759 Add trivial function to compare GstNetAddress. See #520626.
13760 API: GstNetBuffer::gst_netaddress_equal
13762 2008-03-07 16:10:51 +0000 Wim Taymans <wim.taymans@gmail.com>
13764 gst/tcp/gstmultifdsink.c: Update mode property docs, it's deprecated now.
13765 Original commit message from CVS:
13766 * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init):
13767 Update mode property docs, it's deprecated now.
13769 2008-03-07 15:48:51 +0000 Wim Taymans <wim.taymans@gmail.com>
13771 gst/: Remove GstPollMode from gstpoll constructor.
13772 Original commit message from CVS:
13773 * gst-libs/gst/rtsp/gstrtspconnection.c:
13774 (gst_rtsp_connection_create):
13775 * gst/tcp/gstmultifdsink.c: (gst_fdset_mode_get_type),
13776 (gst_multi_fd_sink_class_init), (gst_multi_fd_sink_start):
13777 * gst/tcp/gstmultifdsink.h:
13778 * gst/tcp/gsttcpclientsrc.c: (gst_tcp_client_src_start):
13779 * gst/tcp/gsttcpserversrc.c: (gst_tcp_server_src_start):
13780 Remove GstPollMode from gstpoll constructor.
13782 2008-03-04 00:26:46 +0000 Jan Schmidt <thaytan@mad.scientist.com>
13784 0.10.17.2 pre-release
13785 Original commit message from CVS:
13787 * win32/common/config.h:
13788 0.10.17.2 pre-release
13790 2008-03-03 23:59:45 +0000 Jan Schmidt <thaytan@mad.scientist.com>
13792 gst/Makefile.am: GST_PLUGINS_ALL correctly lists subparse and tcp now, don't distclean them twice
13793 Original commit message from CVS:
13795 GST_PLUGINS_ALL correctly lists subparse and tcp now, don't distclean
13797 * win32/common/libgstinterfaces.def:
13798 * win32/common/libgstrtp.def:
13799 Add new API to the defs
13801 2008-03-03 16:11:50 +0000 Mersad Jelacic <mersad@axis.com>
13803 gst-libs/gst/rtp/gstbasertpaudiopayload.*: API: add gst_base_rtp_audio_payload_set_samplebits_options() to make it po...
13804 Original commit message from CVS:
13805 Patch by: Mersad Jelacic <mersad at axis dot com>
13806 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
13807 * gst-libs/gst/rtp/gstbasertpaudiopayload.h:
13808 API: add gst_base_rtp_audio_payload_set_samplebits_options() to make it
13809 possible to specify the sample size in bits. (#509637)
13811 2008-03-03 13:59:19 +0000 Tim-Philipp Müller <tim@centricular.net>
13813 tests/check/libs/mixer.c: Add a few simple checks for the new message types.
13814 Original commit message from CVS:
13815 * tests/check/libs/mixer.c:
13816 Add a few simple checks for the new message types.
13818 2008-03-03 13:56:38 +0000 Tim-Philipp Müller <tim@centricular.net>
13820 API: add gst_mixer_options_list_changed(), gst_mixer_mixer_changed() and gst_mixer_message_parse_options_list_changed...
13821 Original commit message from CVS:
13822 * docs/libs/gst-plugins-base-libs-sections.txt:
13823 * gst-libs/gst/interfaces/mixer.c: (gst_mixer_option_changed),
13824 (gst_mixer_options_list_changed), (gst_mixer_mixer_changed),
13825 (gst_mixer_message_get_type),
13826 (gst_mixer_message_parse_option_changed),
13827 (gst_mixer_message_parse_options_list_changed):
13828 * gst-libs/gst/interfaces/mixer.h: (GstMixerType),
13829 (GST_MIXER_MESSAGE_OPTION_CHANGED),
13830 (GST_MIXER_MESSAGE_OPTIONS_LIST_CHANGED),
13831 (GST_MIXER_MESSAGE_MIXER_CHANGED):
13832 API: add gst_mixer_options_list_changed(), gst_mixer_mixer_changed()
13833 and gst_mixer_message_parse_options_list_changed(). Fixes #519916.
13835 2008-03-03 13:50:18 +0000 Tim-Philipp Müller <tim@centricular.net>
13837 gst-libs/gst/interfaces/mixeroptions.*: API: add GstMixerOptions::get_values vfunc (#519906)
13838 Original commit message from CVS:
13839 * gst-libs/gst/interfaces/mixeroptions.c: (gst_mixer_options_init),
13840 (gst_mixer_options_get_values):
13841 * gst-libs/gst/interfaces/mixeroptions.h:
13842 (GST_MIXER_OPTIONS_GET_CLASS), (GstMixerOptionsClass),
13843 (_GstMixerOptions), (_GstMixerOptionsClass):
13844 API: add GstMixerOptions::get_values vfunc (#519906)
13846 2008-03-03 12:01:15 +0000 Peter Kjellerstedt <pkj@axis.com>
13848 configure.ac: Use AG_GST_CHECK_PLUGIN and AG_GST_DISABLE_PLUGIN to simplify which plug-ins are included/excluded. (#4...
13849 Original commit message from CVS:
13851 Use AG_GST_CHECK_PLUGIN and AG_GST_DISABLE_PLUGIN to simplify which
13852 plug-ins are included/excluded. (#498222)
13854 2008-03-03 06:22:39 +0000 Sebastian Dröge <slomo@circular-chaos.org>
13856 gst/typefind/gsttypefindfunctions.c: Add typefinder for IMelody files, using audio/x-imelody.
13857 Original commit message from CVS:
13858 * gst/typefind/gsttypefindfunctions.c: (plugin_init):
13859 Add typefinder for IMelody files, using audio/x-imelody.
13862 2008-03-03 06:04:31 +0000 Sebastian Dröge <slomo@circular-chaos.org>
13864 Correct all relevant warnings found by the sparse semantic code analyzer. This include marking several symbols static...
13865 Original commit message from CVS:
13866 * ext/alsa/gstalsamixertrack.c: (gst_alsa_mixer_track_get_type):
13867 * ext/alsa/gstalsasink.c: (set_hwparams):
13868 * ext/alsa/gstalsasrc.c: (set_hwparams):
13869 * ext/gio/gstgio.c: (gst_gio_uri_handler_get_uri):
13870 * ext/ogg/gstoggmux.h:
13871 * ext/ogg/gstogmparse.c:
13872 * gst-libs/gst/audio/audio.c:
13873 * gst-libs/gst/fft/kiss_fft_f64.c: (kiss_fft_f64_alloc):
13874 * gst-libs/gst/pbutils/missing-plugins.c:
13875 (gst_missing_uri_sink_message_new),
13876 (gst_missing_element_message_new),
13877 (gst_missing_decoder_message_new),
13878 (gst_missing_encoder_message_new):
13879 * gst-libs/gst/rtp/gstbasertppayload.c:
13880 * gst-libs/gst/rtp/gstrtcpbuffer.c:
13881 (gst_rtcp_packet_bye_get_reason):
13882 * gst/audioconvert/gstaudioconvert.c:
13883 * gst/audioresample/gstaudioresample.c:
13884 * gst/ffmpegcolorspace/imgconvert.c:
13885 * gst/playback/test.c: (gen_video_element), (gen_audio_element):
13886 * gst/typefind/gsttypefindfunctions.c:
13887 * gst/videoscale/vs_4tap.c:
13888 * gst/videoscale/vs_4tap.h:
13889 * sys/v4l/gstv4lelement.c:
13890 * sys/v4l/gstv4lsrc.c: (gst_v4lsrc_get_any_caps):
13891 * sys/v4l/v4l_calls.c:
13892 * sys/v4l/v4lsrc_calls.c: (gst_v4lsrc_capture_init),
13893 (gst_v4lsrc_try_capture):
13894 * sys/ximage/ximagesink.c: (gst_ximagesink_check_xshm_calls),
13895 (gst_ximagesink_ximage_new):
13896 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_check_xshm_calls),
13897 (gst_xvimagesink_xvimage_new):
13898 * tests/check/elements/audioconvert.c:
13899 * tests/check/elements/audioresample.c:
13900 (fail_unless_perfect_stream):
13901 * tests/check/elements/audiotestsrc.c: (setup_audiotestsrc):
13902 * tests/check/elements/decodebin.c:
13903 * tests/check/elements/gdpdepay.c: (setup_gdpdepay),
13904 (setup_gdpdepay_streamheader):
13905 * tests/check/elements/gdppay.c: (setup_gdppay), (GST_START_TEST),
13906 (setup_gdppay_streamheader):
13907 * tests/check/elements/gnomevfssink.c: (setup_gnomevfssink):
13908 * tests/check/elements/multifdsink.c: (setup_multifdsink):
13909 * tests/check/elements/textoverlay.c:
13910 * tests/check/elements/videorate.c: (setup_videorate):
13911 * tests/check/elements/videotestsrc.c: (setup_videotestsrc):
13912 * tests/check/elements/volume.c: (setup_volume):
13913 * tests/check/elements/vorbisdec.c: (setup_vorbisdec):
13914 * tests/check/elements/vorbistag.c:
13915 * tests/check/generic/clock-selection.c:
13916 * tests/check/generic/states.c: (setup), (teardown):
13917 * tests/check/libs/cddabasesrc.c:
13918 * tests/check/libs/video.c:
13919 * tests/check/pipelines/gio.c:
13920 * tests/check/pipelines/oggmux.c:
13921 * tests/check/pipelines/simple-launch-lines.c:
13922 (simple_launch_lines_suite):
13923 * tests/check/pipelines/streamheader.c:
13924 * tests/check/pipelines/theoraenc.c:
13925 * tests/check/pipelines/vorbisdec.c:
13926 * tests/check/pipelines/vorbisenc.c:
13927 * tests/examples/seek/scrubby.c:
13928 * tests/examples/seek/seek.c: (query_positions_elems),
13929 (query_positions_pads):
13930 * tests/icles/stress-xoverlay.c: (myclock):
13931 Correct all relevant warnings found by the sparse semantic code
13932 analyzer. This include marking several symbols static, using
13933 NULL instead of 0 for pointers and using "foo (void)" instead
13934 of "foo ()" for declarations.
13935 * win32/common/libgstrtp.def:
13936 Add gst_rtp_buffer_set_extension_data to the symbol definition file.
13938 2008-03-02 18:43:15 +0000 José Alburquerque <jaalburqu@svn.gnome.org>
13940 gst/playback/gstplaybin2.c: Make the function signature of the _get_*_tags() functions match the signature of the vfu...
13941 Original commit message from CVS:
13942 Patch by: José Alburquerque <jaalburqu svn gnome org>
13943 * gst/playback/gstplaybin2.c:
13944 Make the function signature of the _get_*_tags() functions match
13945 the signature of the vfuncs they implement, ie. return a
13946 GstTagList rather than a GstStructure, which is more correct,
13947 even if one is typedef'ed to the other (#518940).
13949 2008-03-02 18:32:36 +0000 Tim-Philipp Müller <tim@centricular.net>
13951 gst-libs/gst/rtsp/gstrtspconnection.c: Don't include unix headers unconditionally (fixes #518037).
13952 Original commit message from CVS:
13953 * gst-libs/gst/rtsp/gstrtspconnection.c:
13954 Don't include unix headers unconditionally (fixes #518037).
13956 2008-03-02 18:24:37 +0000 Tim-Philipp Müller <tim@centricular.net>
13958 tests/check/libs/video.c: Add unit test that makes sure that the strides, offsets and sizes returned for the various ...
13959 Original commit message from CVS:
13960 * tests/check/libs/video.c: (paintinfo), (paintinfo_struct),
13961 (fourcc_list_struct), (fourcc_list), (fourcc_get_size),
13962 (paint_setup_I420), (paint_setup_YV12), (paint_setup_AYUV),
13963 (paint_setup_YUY2), (paint_setup_UYVY), (paint_setup_YVYU),
13964 (paint_setup_IYU2), (paint_setup_Y41B), (paint_setup_Y42B),
13965 (paint_setup_Y800), (paint_setup_YVU9), (paint_setup_YUV9),
13966 (gst_video_format_is_packed), (video_format_is_packed):
13967 Add unit test that makes sure that the strides, offsets and
13968 sizes returned for the various YUV formats by the new video API
13969 match the old reference implementation in videotestsrc.
13971 2008-03-02 18:20:44 +0000 Tim-Philipp Müller <tim@centricular.net>
13973 gst-libs/gst/video/video.*: API: add GST_VIDEO_FORMAT_Y41B and GST_VIDEO_FORMAT_Y42B.
13974 Original commit message from CVS:
13975 * gst-libs/gst/video/video.c: (gst_video_calculate_display_ratio),
13976 (gst_video_format_from_fourcc), (gst_video_format_to_fourcc),
13977 (gst_video_format_is_rgb), (gst_video_format_is_yuv),
13978 (gst_video_format_has_alpha), (gst_video_format_get_row_stride),
13979 (gst_video_format_get_pixel_stride),
13980 (gst_video_format_get_component_width),
13981 (gst_video_format_get_component_height),
13982 (gst_video_format_get_component_offset), (gst_video_format_get_size):
13983 * gst-libs/gst/video/video.h: (GST_VIDEO_FORMAT_Y41B),
13984 (GST_VIDEO_FORMAT_Y42B):
13985 API: add GST_VIDEO_FORMAT_Y41B and GST_VIDEO_FORMAT_Y42B.
13987 2008-03-02 18:07:10 +0000 Tim-Philipp Müller <tim@centricular.net>
13989 gst-libs/gst/video/video.c: YV12 is I420 with swapped components 1 and 2, so the offset of component 1 for I420 shoul...
13990 Original commit message from CVS:
13991 * gst-libs/gst/video/video.c: (gst_video_format_get_component_offset):
13992 YV12 is I420 with swapped components 1 and 2, so the offset of
13993 component 1 for I420 should be the offset for component 2 for YV12
13996 2008-02-29 21:48:00 +0000 Rene Stadler <mail@renestadler.de>
13998 sys/v4l/gstv4lelement.c: Add missing semicolon to fix indentation.
13999 Original commit message from CVS:
14000 * sys/v4l/gstv4lelement.c:
14001 Add missing semicolon to fix indentation.
14003 2008-02-29 18:44:36 +0000 Julien Moutte <julien@moutte.net>
14005 ext/alsa/gstalsa.c: Probe for IEC958 pcm to detect if we can do SPDIF output.
14006 Original commit message from CVS:
14007 2008-02-29 Julien Moutte <julien@fluendo.com>
14008 * ext/alsa/gstalsa.c: (gst_alsa_open_iec958_pcm),
14009 (gst_alsa_probe_supported_formats): Probe for IEC958 pcm to
14011 if we can do SPDIF output.
14012 * ext/alsa/gstalsa.h:
14013 * ext/alsa/gstalsasink.c: (set_hwparams), (alsasink_parse_spec),
14014 (gst_alsasink_prepare), (gst_alsasink_close),
14015 (gst_alsasink_write):
14016 * ext/alsa/gstalsasink.h: Initial support for SPDIF.
14017 * gst-libs/gst/audio/gstringbuffer.c:
14018 (gst_ring_buffer_parse_caps):
14019 * gst-libs/gst/audio/gstringbuffer.h: Add non linear buffer
14021 to support AC3, EC3 and IEC958 buffers.
14023 2008-02-29 17:59:16 +0000 Tim-Philipp Müller <tim@centricular.net>
14025 gst-libs/gst/interfaces/mixer.c: De-cruft and fix message type assertions (NULL is not a really valid mixer message t...
14026 Original commit message from CVS:
14027 * gst-libs/gst/interfaces/mixer.c: (GST_MIXER_MESSAGE_HAS_TYPE),
14028 (gst_mixer_message_parse_mute_toggled),
14029 (gst_mixer_message_parse_record_toggled),
14030 (gst_mixer_message_parse_volume_changed),
14031 (gst_mixer_message_parse_option_changed):
14032 De-cruft and fix message type assertions (NULL is not a really
14033 valid mixer message type string).
14035 2008-02-29 14:52:02 +0000 Wim Taymans <wim.taymans@gmail.com>
14037 ext/libvisual/visual.c: When negotiating, actually start from a format that we can support instead of from the too ge...
14038 Original commit message from CVS:
14039 * ext/libvisual/visual.c: (gst_vis_src_negotiate):
14040 When negotiating, actually start from a format that we can support
14041 instead of from the too generic template.
14043 2008-02-29 12:26:48 +0000 Wim Taymans <wim.taymans@gmail.com>
14045 gst/playback/gstplaybin2.c: Enable vis setting.
14046 Original commit message from CVS:
14047 * gst/playback/gstplaybin2.c: (gst_play_bin_set_property):
14048 Enable vis setting.
14049 * gst/playback/gstplaysink.c: (gst_play_sink_init),
14050 (gst_play_sink_dispose), (gst_play_sink_vis_unblocked),
14051 (gst_play_sink_vis_blocked), (gst_play_sink_set_vis_plugin),
14053 Implement vis switching while playing.
14055 2008-02-29 00:04:57 +0000 David Schleef <ds@schleef.org>
14057 gst-libs/gst/riff/riff-media.c: Add Dirac mapping
14058 Original commit message from CVS:
14059 * gst-libs/gst/riff/riff-media.c: Add Dirac mapping
14061 2008-02-28 10:54:14 +0000 Peter Kjellerstedt <pkj@axis.com>
14063 gst/tcp/: Removed fdset and stress test, they are now known as GstPoll in core.
14064 Original commit message from CVS:
14065 Patch by: Peter Kjellerstedt <pkj at axis com>
14066 * gst/tcp/Makefile.am:
14067 * gst/tcp/fdsetstress.c:
14068 * gst/tcp/gstfdset.c:
14069 * gst/tcp/gstfdset.h:
14070 Removed fdset and stress test, they are now known as GstPoll in
14072 * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init),
14073 (gst_multi_fd_sink_add_full), (gst_multi_fd_sink_remove),
14074 (gst_multi_fd_sink_clear), (gst_multi_fd_sink_remove_client_link),
14075 (gst_multi_fd_sink_handle_client_write),
14076 (gst_multi_fd_sink_queue_buffer),
14077 (gst_multi_fd_sink_handle_clients), (gst_multi_fd_sink_start),
14078 (gst_multi_fd_sink_stop):
14079 * gst/tcp/gstmultifdsink.h:
14080 * gst/tcp/gsttcp.c: (gst_tcp_socket_read), (gst_tcp_socket_close),
14081 (gst_tcp_read_buffer), (gst_tcp_gdp_read_buffer),
14082 (gst_tcp_gdp_read_caps):
14083 * gst/tcp/gsttcp.h:
14084 * gst/tcp/gsttcpclientsink.c: (gst_tcp_client_sink_init),
14085 (gst_tcp_client_sink_setcaps), (gst_tcp_client_sink_render),
14086 (gst_tcp_client_sink_start), (gst_tcp_client_sink_stop):
14087 * gst/tcp/gsttcpclientsink.h:
14088 * gst/tcp/gsttcpclientsrc.c: (gst_tcp_client_src_init),
14089 (gst_tcp_client_src_create), (gst_tcp_client_src_start),
14090 (gst_tcp_client_src_stop), (gst_tcp_client_src_unlock):
14091 * gst/tcp/gsttcpclientsrc.h:
14092 * gst/tcp/gsttcpserversink.c: (gst_tcp_server_sink_handle_wait),
14093 (gst_tcp_server_sink_init_send), (gst_tcp_server_sink_close):
14094 * gst/tcp/gsttcpserversink.h:
14095 * gst/tcp/gsttcpserversrc.c: (gst_tcp_server_src_init),
14096 (gst_tcp_server_src_create), (gst_tcp_server_src_start),
14097 (gst_tcp_server_src_stop), (gst_tcp_server_src_unlock):
14098 * gst/tcp/gsttcpserversrc.h:
14099 Port to GstPoll. See #505417.
14101 2008-02-28 09:54:14 +0000 Wim Taymans <wim.taymans@gmail.com>
14104 Patch Changelog a bit to give credit and refer to the relevant bug.
14105 Original commit message from CVS:
14106 Patch Changelog a bit to give credit and refer to the
14109 2008-02-28 09:50:52 +0000 Wim Taymans <wim.taymans@gmail.com>
14111 gst-libs/gst/rtsp/gstrtspconnection.*: Use GstPoll for the rtsp connection.
14112 Original commit message from CVS:
14113 * gst-libs/gst/rtsp/gstrtspconnection.c:
14114 (gst_rtsp_connection_create), (gst_rtsp_connection_connect),
14115 (gst_rtsp_connection_write), (gst_rtsp_connection_read_internal),
14116 (gst_rtsp_connection_receive), (gst_rtsp_connection_close),
14117 (gst_rtsp_connection_free), (gst_rtsp_connection_poll),
14118 (gst_rtsp_connection_flush):
14119 * gst-libs/gst/rtsp/gstrtspconnection.h:
14120 Use GstPoll for the rtsp connection.
14122 2008-02-27 12:19:31 +0000 Wim Taymans <wim.taymans@gmail.com>
14124 tests/examples/seek/seek.c: Add combo box for visualisations, populate it with a factory list of all visualisation pl...
14125 Original commit message from CVS:
14126 * tests/examples/seek/seek.c: (vis_toggle_cb), (filter_features),
14127 (init_visualization_features), (vis_combo_cb), (shot_cb), (main):
14128 Add combo box for visualisations, populate it with a factory list
14129 of all visualisation plugins, configure vis plugin instance in
14132 2008-02-27 10:55:03 +0000 Wim Taymans <wim.taymans@gmail.com>
14134 tests/check/libs/rtp.c: Add check for RTP buffer defaults, padding and marker bit API.
14135 Original commit message from CVS:
14136 * tests/check/libs/rtp.c: (GST_START_TEST):
14137 Add check for RTP buffer defaults, padding and marker bit API.
14139 2008-02-27 10:42:08 +0000 Sebastian Dröge <slomo@circular-chaos.org>
14141 gst-libs/gst/cdda/sha1.c: Use memcpy() instead of upcasting a byte array to long *. This fixes an unaligned memory ac...
14142 Original commit message from CVS:
14143 * gst-libs/gst/cdda/sha1.c: (sha_transform):
14144 Use memcpy() instead of upcasting a byte array to long *. This
14145 fixes an unaligned memory access, resulting in SIGBUS on IA64.
14146 This should be ported to GCheckSum once we can use GLib 2.16.
14147 Partially fixes bug #500833.
14149 2008-02-27 10:23:27 +0000 Tim-Philipp Müller <tim@centricular.net>
14151 gst-libs/gst/tag/gsttagdemux.c: Push tag event after the newsegment event. Log the pointer of the buffer we're actual...
14152 Original commit message from CVS:
14153 * gst-libs/gst/tag/gsttagdemux.c: (gst_tag_demux_chain):
14154 Push tag event after the newsegment event. Log the pointer of
14155 the buffer we're actually going to push rather than the buffer
14156 we're feeding to _make_metadata_writable().
14158 2008-02-25 07:21:33 +0000 Sebastian Dröge <slomo@circular-chaos.org>
14160 gst/typefind/gsttypefindfunctions.c: Comment smoke typefinder for now. The smokedec plugin needs one frame per buffer...
14161 Original commit message from CVS:
14162 * gst/typefind/gsttypefindfunctions.c: (plugin_init):
14163 Comment smoke typefinder for now. The smokedec plugin needs one
14164 frame per buffer but we have no parser yet, thus it simply crashes
14165 in most situations.
14167 2008-02-25 06:48:14 +0000 Sebastian Dröge <slomo@circular-chaos.org>
14169 gst/typefind/gsttypefindfunctions.c: Add typefinder for the smoke video codec. Copied from the jpeg plugin.
14170 Original commit message from CVS:
14171 * gst/typefind/gsttypefindfunctions.c: (plugin_init):
14172 Add typefinder for the smoke video codec. Copied from the jpeg plugin.
14174 2008-02-25 06:29:09 +0000 Sebastian Dröge <slomo@circular-chaos.org>
14176 gst/typefind/gsttypefindfunctions.c: Add midi typefinder, copied from the timidity plugin.
14177 Original commit message from CVS:
14178 * gst/typefind/gsttypefindfunctions.c: (mid_type_find),
14180 Add midi typefinder, copied from the timidity plugin.
14182 2008-02-23 09:51:26 +0000 Tomasz Sałaciński <tsalacinski@gmail.com>
14184 Forward slashes at the beginning and end of a line also signify italics (Fixes: #518162).
14185 Original commit message from CVS:
14186 Based on patch by: Tomasz Sałaciński <tsalacinski gmail com>
14187 * gst/subparse/gstsubparse.c: (parse_mdvdsub):
14188 * tests/check/elements/subparse.c: (test_microdvd_with_italics),
14190 Forward slashes at the beginning and end of a line also signify
14191 italics (Fixes: #518162).
14193 2008-02-22 06:38:08 +0000 Sebastian Dröge <slomo@circular-chaos.org>
14195 tests/check/gst-plugins-base.supp: Add a suppression for a cached value in GIO that wasn't moved while moving gio fro...
14196 Original commit message from CVS:
14197 * tests/check/gst-plugins-base.supp:
14198 Add a suppression for a cached value in GIO that wasn't moved
14199 while moving gio from -bad to -base.
14201 2008-02-22 05:27:24 +0000 Brian Cameron <brian.cameron@sun.com>
14203 configure.ac: Don't hardcode -Wall and -Werror for configure checks, this fails with non-GCC compilers. Fixes bug #51...
14204 Original commit message from CVS:
14205 Patch by: Brian Cameron <brian dot cameron at sun dot com>
14207 Don't hardcode -Wall and -Werror for configure checks, this fails
14208 with non-GCC compilers. Fixes bug #517991.
14210 2008-02-21 08:05:10 +0000 Stefan Kost <ensonic@users.sourceforge.net>
14212 gst/audiotestsrc/gstaudiotestsrc.c: Mark buffers as GAP,if volume is 0.0 and fix the previous logic.
14213 Original commit message from CVS:
14214 * gst/audiotestsrc/gstaudiotestsrc.c:
14215 Mark buffers as GAP,if volume is 0.0 and fix the previous logic.
14217 2008-02-20 15:37:36 +0000 Sebastian Dröge <slomo@circular-chaos.org>
14219 ext/gnomevfs/gstgnomevfssink.c: Return FALSE when seeking for a new segment fails instead of silently ignoring the fa...
14220 Original commit message from CVS:
14221 * ext/gnomevfs/gstgnomevfssink.c:
14222 (gst_gnome_vfs_sink_handle_event):
14223 Return FALSE when seeking for a new segment fails instead
14224 of silently ignoring the failure and appending every buffer
14225 that comes for the new segment.
14227 2008-02-20 11:52:28 +0000 Wim Taymans <wim.taymans@gmail.com>
14229 gst/playback/gstplaysink.c: Recursively search the sink element for a last-frame property so that we can also find th...
14230 Original commit message from CVS:
14231 * gst/playback/gstplaysink.c: (find_property),
14232 (gst_play_sink_find_property), (gen_video_chain),
14233 (gst_play_sink_reconfigure), (gst_play_sink_get_last_frame):
14234 Recursively search the sink element for a last-frame property so that we
14235 can also find the property in autovideosink and friends that don't
14236 always proxy the internal sink properties.
14238 2008-02-19 20:42:09 +0000 Tim-Philipp Müller <tim@centricular.net>
14240 gst-libs/gst/audio/multichannel.c: Fix confusing terminology in docs and code: structure fields are 'fields' and not ...
14241 Original commit message from CVS:
14242 * gst-libs/gst/audio/multichannel.c:
14243 (GST_AUDIO_CHANNEL_POSITIONS_FIELD_NAME),
14244 (gst_audio_get_channel_positions), (gst_audio_set_channel_positions),
14245 (gst_audio_set_structure_channel_positions_list),
14246 (add_list_to_struct), (gst_audio_set_caps_channel_positions_list),
14247 (gst_audio_fixate_channel_positions):
14248 Fix confusing terminology in docs and code: structure fields are
14249 'fields' and not 'properties'.
14251 2008-02-19 20:36:58 +0000 Tim-Philipp Müller <tim@centricular.net>
14253 gst-libs/gst/audio/multichannel.c: Give more useful warning messages if one of the channel layout enums passed to us ...
14254 Original commit message from CVS:
14255 * gst-libs/gst/audio/multichannel.c:
14256 (gst_audio_check_channel_positions), (add_list_to_struct):
14257 Give more useful warning messages if one of the channel
14258 layout enums passed to us is invalid and if the "channels"
14259 field in the caps has a GType we don't expect.
14261 2008-02-19 20:22:09 +0000 Tim-Philipp Müller <tim@centricular.net>
14263 gst-libs/gst/audio/multichannel.c: Fix typo in docs blurb.
14264 Original commit message from CVS:
14265 * gst-libs/gst/audio/multichannel.c:
14266 Fix typo in docs blurb.
14268 2008-02-19 16:16:55 +0000 Josep Torra Valles <josep@fluendo.com>
14270 gst/typefind/gsttypefindfunctions.c: Increase the MPEG PS typefind lookup to fix typefinding on HD clips.
14271 Original commit message from CVS:
14272 2008-02-19 Julien Moutte <julien@fluendo.com>
14273 Patch by: Josep Torra Valles <josep@fluendo.com>
14274 * gst/typefind/gsttypefindfunctions.c: Increase the MPEG PS
14275 typefind lookup to fix typefinding on HD clips.
14277 2008-02-19 15:50:37 +0000 Tim-Philipp Müller <tim@centricular.net>
14279 gst/playback/gstscreenshot.*: Fix up copyright (I rewrote the GStreamer-0.10 code for this from scratch back in the d...
14280 Original commit message from CVS:
14281 * gst/playback/gstscreenshot.c:
14282 * gst/playback/gstscreenshot.h:
14283 Fix up copyright (I rewrote the GStreamer-0.10 code for
14284 this from scratch back in the days).
14286 2008-02-19 15:02:33 +0000 Wim Taymans <wim.taymans@gmail.com>
14288 gst/playback/: Add screenshot conversion code from totem.
14289 Original commit message from CVS:
14290 * gst/playback/Makefile.am:
14291 * gst/playback/gstscreenshot.c: (feed_fakesrc), (save_result),
14292 (create_element), (gst_play_frame_conv_convert):
14293 * gst/playback/gstscreenshot.h:
14294 Add screenshot conversion code from totem.
14295 * gst/playback/gstplay-marshal.list:
14296 * gst/playback/gstplaybin2.c: (gst_play_marshal_BUFFER__BOXED),
14297 (gst_play_bin_class_init), (gst_play_bin_convert_frame),
14298 (gst_play_bin_get_property), (no_more_pads_cb), (activate_group):
14299 Implement frame property to get a color-unconverted snapshot.
14300 Implement convert-frame action signal to get a converted snapshot image.
14301 Configure connection speed in uridecodebin.
14302 Document some more properties.
14303 * gst/playback/gstplaysink.c: (gst_play_sink_class_init),
14304 (gen_video_chain), (gen_audio_chain), (gst_play_sink_reconfigure),
14305 (gst_play_sink_get_last_frame):
14306 * gst/playback/gstplaysink.h:
14307 Use last-buffer property of the video sink to get a video snapshot.
14308 * tests/examples/seek/seek.c: (shot_cb), (main):
14309 Add snapshot button for playbin2 and use the frame property to save the
14310 frame as a png in the current directory.
14312 2008-02-19 11:45:56 +0000 Josep Torra Valles <josep@fluendo.com>
14314 gst/typefind/gsttypefindfunctions.c: Add typefinding support for h264 elementary streams.
14315 Original commit message from CVS:
14316 Patch by: Josep Torra Valles <josep at fluendo dot com>
14317 * gst/typefind/gsttypefindfunctions.c: (h264_video_type_find),
14319 Add typefinding support for h264 elementary streams.
14322 2008-02-18 13:51:34 +0000 Stefan Kost <ensonic@users.sourceforge.net>
14324 configure.ac: Require CVS of core for new API in collectpads.
14325 Original commit message from CVS:
14327 Require CVS of core for new API in collectpads.
14328 * gst/adder/gstadder.c:
14329 Use new API to make adder sparse stream aware.
14331 2008-02-18 11:54:15 +0000 Wim Taymans <wim.taymans@gmail.com>
14333 gst/playback/gstplaybin2.c: Get the object data correct so that we can remove our channels correctly.
14334 Original commit message from CVS:
14335 * gst/playback/gstplaybin2.c: (pad_added_cb), (pad_removed_cb),
14337 Get the object data correct so that we can remove our channels
14339 * gst/playback/gstplaysink.c: (gen_video_chain), (gen_audio_chain),
14340 (gen_vis_chain), (gst_play_sink_reconfigure),
14341 (gst_play_sink_request_pad):
14342 Add option to disable async behaviour in the sinks when possible. This
14343 makes it possible to avoid an audio queue when dealing with
14345 Add option to add a queue for the audio path.
14346 * tests/examples/seek/seek.c: (clear_streams), (update_streams),
14348 Disable the vis checkbox to match the defaults of playbin2.
14349 Only get the stream info when we need to.
14351 2008-02-17 05:15:45 +0000 Sebastian Dröge <slomo@circular-chaos.org>
14353 ext/gio/: Don't use async operations as they require a running main loop.
14354 Original commit message from CVS:
14355 * ext/gio/gstgiobasesink.c: (gst_gio_base_sink_stop),
14356 (gst_gio_base_sink_set_stream):
14357 * ext/gio/gstgiobasesrc.c: (gst_gio_base_src_stop),
14358 (gst_gio_base_src_set_stream):
14359 * ext/gio/gstgiosink.c: (gst_gio_sink_start):
14360 * ext/gio/gstgiosrc.c: (gst_gio_src_start):
14361 Don't use async operations as they require a running main loop.
14362 This makes us block again when closing streams and unable
14363 to mount the enclosing volume of an URI if it isn't yet.
14365 2008-02-15 18:38:52 +0000 Wim Taymans <wim.taymans@gmail.com>
14367 gst/playback/gstplaysink.c: Move tee in front of the audio and vis pipelines.
14368 Original commit message from CVS:
14369 * gst/playback/gstplaysink.c: (gst_play_sink_set_mute),
14370 (gst_play_sink_get_mute), (gen_video_chain), (gen_audio_chain),
14371 (gen_vis_chain), (gst_play_sink_reconfigure),
14372 (gst_play_sink_request_pad):
14373 Move tee in front of the audio and vis pipelines.
14374 Add queue for audio for now.
14375 Add visualisation support.
14376 * tests/examples/seek/seek.c: (main):
14377 Visualisation is by default disabled.
14379 2008-02-15 11:58:06 +0000 Sebastian Dröge <slomo@circular-chaos.org>
14381 ext/gio/: Improve debugging a bit.
14382 Original commit message from CVS:
14383 * ext/gio/gstgiobasesink.c: (close_stream_cb):
14384 * ext/gio/gstgiobasesrc.c: (close_stream_cb):
14385 Improve debugging a bit.
14386 * ext/gio/gstgiosink.c: (mount_cb), (gst_gio_sink_start):
14387 * ext/gio/gstgiosink.h:
14388 * ext/gio/gstgiosrc.c: (mount_cb), (gst_gio_src_start):
14389 * ext/gio/gstgiosrc.h:
14390 Try to mount the enclosing volume of a GFile if it isn't mounted
14391 yet. This requires us to wait for an async operation to finish, done
14392 with an nested GMainLoop. Authentication is not supported yet, will
14395 2008-02-14 18:24:42 +0000 Wim Taymans <wim.taymans@gmail.com>
14397 gst/playback/: Add mute property.
14398 Original commit message from CVS:
14399 * gst/playback/gstplaybin2.c: (gst_play_bin_class_init),
14400 (gst_play_bin_set_property), (gst_play_bin_get_property),
14401 (pad_added_cb), (pad_removed_cb), (no_more_pads_cb):
14402 * gst/playback/gstplaysink.c: (gst_play_sink_set_mute),
14403 (gst_play_sink_get_mute), (gen_audio_chain):
14404 * gst/playback/gstplaysink.h:
14406 * gst/playback/gststreamselector.c: (gst_selector_pad_event),
14407 (gst_selector_pad_chain):
14408 * gst/playback/gststreamselector.h:
14409 Make sure we forward the event only once.
14410 * tests/examples/seek/seek.c: (stop_cb), (mute_toggle_cb), (main):
14411 Add and implement the mute button for playbin2.
14413 2008-02-13 14:34:55 +0000 Tommi Myöhänen <ext-tommi.myohanen@nokia.com>
14415 ext/alsa/gstalsasink.c: Add some more debug info.
14416 Original commit message from CVS:
14417 Patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com>
14418 * ext/alsa/gstalsasink.c: (set_hwparams), (gst_alsasink_delay):
14419 Add some more debug info.
14420 Make sure we never return a negative delay. Fixes #516246.
14422 2008-02-12 20:09:07 +0000 Tim-Philipp Müller <tim@centricular.net>
14424 ext/alsa/gstalsasink.c: Revert patch that makes the sink hold the object lock when calling snd_pcm_delay(), since it ...
14425 Original commit message from CVS:
14426 * ext/alsa/gstalsasink.c: (gst_alsasink_delay):
14427 Revert patch that makes the sink hold the object lock when
14428 calling snd_pcm_delay(), since it breaks playback for me.
14430 2008-02-12 19:50:36 +0000 Julien Moutte <julien@moutte.net>
14432 tests/examples/seek/seek.c: Add some seek flags when changing rate.
14433 Original commit message from CVS:
14434 2008-02-12 Julien Moutte <julien@fluendo.com>
14435 * tests/examples/seek/seek.c: (rate_spinbutton_changed_cb): Add
14436 some seek flags when changing rate.
14438 2008-02-12 14:51:26 +0000 Wim Taymans <wim.taymans@gmail.com>
14440 gst-libs/gst/rtp/gstbasertpaudiopayload.c: Fix potential leaks.
14441 Original commit message from CVS:
14442 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
14443 (gst_base_rtp_audio_payload_handle_frame_based_buffer),
14444 (gst_base_rtp_audio_payload_handle_sample_based_buffer):
14445 Fix potential leaks.
14446 * gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_chain):
14447 Fix leak when there is no function configured.
14449 2008-02-12 11:36:27 +0000 Sebastian Dröge <slomo@circular-chaos.org>
14451 sys/v4l/v4lsrc_calls.c: Correctly chain up the finalize method.
14452 Original commit message from CVS:
14453 * sys/v4l/v4lsrc_calls.c: (gst_v4lsrc_buffer_class_init),
14454 (gst_v4lsrc_buffer_finalize):
14455 Correctly chain up the finalize method.
14457 2008-02-12 09:24:11 +0000 Sebastian Dröge <slomo@circular-chaos.org>
14459 ext/gio/: Add documentation and example code for giostreamsink/giostreamsrc.
14460 Original commit message from CVS:
14461 * ext/gio/gstgiostreamsink.c:
14462 * ext/gio/gstgiostreamsrc.c:
14463 Add documentation and example code for giostreamsink/giostreamsrc.
14464 * tests/check/pipelines/gio.c: (GST_START_TEST):
14465 Ask the GMemoryOutputStream for the data instead of assuming that
14466 the pointer to the data stayed the same. It could've been realloc'ed.
14468 2008-02-12 08:55:57 +0000 Sebastian Dröge <slomo@circular-chaos.org>
14470 ext/gio/: Make the documentation of giosink/giosrc complete, large parts are based on the gnomevfssink/gnomevfssrc docs.
14471 Original commit message from CVS:
14472 * ext/gio/gstgiosink.c:
14473 * ext/gio/gstgiosrc.c:
14474 Make the documentation of giosink/giosrc complete, large parts
14475 are based on the gnomevfssink/gnomevfssrc docs.
14477 2008-02-12 08:13:59 +0000 Sebastian Dröge <slomo@circular-chaos.org>
14479 docs/plugins/: Add the GIO documentation again and while at that run make update.
14480 Original commit message from CVS:
14481 * docs/plugins/gst-plugins-base-plugins-docs.sgml:
14482 * docs/plugins/gst-plugins-base-plugins-sections.txt:
14483 * docs/plugins/gst-plugins-base-plugins.args:
14484 * docs/plugins/gst-plugins-base-plugins.hierarchy:
14485 * docs/plugins/gst-plugins-base-plugins.interfaces:
14486 * docs/plugins/gst-plugins-base-plugins.prerequisites:
14487 * docs/plugins/gst-plugins-base-plugins.signals:
14488 * docs/plugins/inspect/plugin-adder.xml:
14489 * docs/plugins/inspect/plugin-audioconvert.xml:
14490 * docs/plugins/inspect/plugin-audiorate.xml:
14491 * docs/plugins/inspect/plugin-audioresample.xml:
14492 * docs/plugins/inspect/plugin-decodebin.xml:
14493 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
14494 * docs/plugins/inspect/plugin-gdp.xml:
14495 * docs/plugins/inspect/plugin-gio.xml:
14496 * docs/plugins/inspect/plugin-gnomevfs.xml:
14497 * docs/plugins/inspect/plugin-libvisual.xml:
14498 * docs/plugins/inspect/plugin-ogg.xml:
14499 * docs/plugins/inspect/plugin-pango.xml:
14500 * docs/plugins/inspect/plugin-playback.xml:
14501 * docs/plugins/inspect/plugin-queue2.xml:
14502 * docs/plugins/inspect/plugin-subparse.xml:
14503 * docs/plugins/inspect/plugin-theora.xml:
14504 * docs/plugins/inspect/plugin-uridecodebin.xml:
14505 * docs/plugins/inspect/plugin-videorate.xml:
14506 * docs/plugins/inspect/plugin-videoscale.xml:
14507 * docs/plugins/inspect/plugin-volume.xml:
14508 * docs/plugins/inspect/plugin-vorbis.xml:
14509 Add the GIO documentation again and while at that run make update.
14511 2008-02-11 20:23:44 +0000 Tim-Philipp Müller <tim@centricular.net>
14513 ext/alsa/: Don't use snd_pcm_sw_params_set_xfer_align() if we're compiling against libasound >= 1.0.16, since it's be...
14514 Original commit message from CVS:
14515 * ext/alsa/gstalsa.h: (GST_CHECK_ALSA_VERSION):
14516 * ext/alsa/gstalsasink.c: (set_swparams):
14517 * ext/alsa/gstalsasrc.c: (set_swparams), (gst_alsasrc_open):
14518 Don't use snd_pcm_sw_params_set_xfer_align() if we're compiling
14519 against libasound >= 1.0.16, since it's been deprecated in
14520 0.10.16, and alignment is always 1 then, apparently. (#512899)
14522 2008-02-11 18:31:43 +0000 Tim-Philipp Müller <tim@centricular.net>
14524 gst/playback/: Handle case where we can't create the volume element a bit better (#514307).
14525 Original commit message from CVS:
14526 * gst/playback/gstplaybin.c: (gen_audio_element):
14527 * gst/playback/gstplaysink.c: (gen_audio_chain):
14528 Handle case where we can't create the volume element a bit
14531 2008-02-11 18:02:13 +0000 Tim-Philipp Müller <tim@centricular.net>
14533 ext/gnomevfs/: Add support for https protocol. Fixes #510229.
14534 Original commit message from CVS:
14535 * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_check_get_range):
14536 * ext/gnomevfs/gstgnomevfsuri.c: (gst_gnomevfs_get_supported_uris):
14537 Add support for https protocol. Fixes #510229.
14539 2008-02-11 17:03:18 +0000 Alan Peevers <peeves@pacbell.net>
14541 ext/alsa/gstalsasink.c: Take appropriate lock when calling alsa methods.
14542 Original commit message from CVS:
14543 2008-02-11 Julien Moutte <julien@fluendo.com>
14544 Patch by: Alan Peevers <peeves@pacbell.net>
14545 * ext/alsa/gstalsasink.c: (gst_alsasink_delay): Take appropriate
14546 lock when calling alsa methods.
14548 2008-02-11 13:03:13 +0000 Tim-Philipp Müller <tim@centricular.net>
14550 gst/typefind/gsttypefindfunctions.c: Bump rank of jpeg and png typefinders, which will return maximum probability in ...
14551 Original commit message from CVS:
14552 * gst/typefind/gsttypefindfunctions.c:
14553 Bump rank of jpeg and png typefinders, which will return maximum
14554 probability in the most common cases (thus short-circuiting more
14555 expensive typefinders like the mp3 one for these two quite common
14558 2008-02-11 09:48:03 +0000 Sebastian Dröge <slomo@circular-chaos.org>
14560 ext/theora/theoraparse.c: Fix long description of the theora parser to be more verbose than just the type name.
14561 Original commit message from CVS:
14562 * ext/theora/theoraparse.c:
14563 Fix long description of the theora parser to be more verbose than just
14566 2008-02-11 06:47:50 +0000 Branko Čibej <brane@xbc.nu>
14568 sys/xvimage/xvimagesink.c: Fix build of xvimagesink if we don't have XShm, e.g. on Mac OS X.
14569 Original commit message from CVS:
14570 Patch by: Branko Čibej <brane at xbc dot nu>
14571 * sys/xvimage/xvimagesink.c:
14572 Fix build of xvimagesink if we don't have XShm, e.g. on Mac OS X.
14575 2008-02-09 10:41:36 +0000 Zaheer Abbas Merali <zaheerabbas@merali.org>
14577 gst/playback/gstplaybasebin.c: Set is_dynamic as True if there are elements with both request and sometimes src pad t...
14578 Original commit message from CVS:
14579 * gst/playback/gstplaybasebin.c:
14580 Set is_dynamic as True if there are elements with both request
14581 and sometimes src pad templates instead of breaking out when it
14582 finds the first pad template that is a src.
14584 2008-02-08 18:17:51 +0000 Wim Taymans <wim.taymans@gmail.com>
14586 tests/examples/seek/seek.c: Add some stream switching and volume gui for playbin2.
14587 Original commit message from CVS:
14588 * tests/examples/seek/seek.c: (stop_cb), (clear_streams),
14589 (update_streams), (video_combo_cb), (audio_combo_cb),
14590 (text_combo_cb), (volume_spinbutton_changed_cb), (main):
14591 Add some stream switching and volume gui for playbin2.
14593 2008-02-08 17:47:37 +0000 Wim Taymans <wim.taymans@gmail.com>
14595 gst/playback/gstplay-marshal.list: Added marshal for streamselector Tags.
14596 Original commit message from CVS:
14597 * gst/playback/gstplay-marshal.list:
14598 Added marshal for streamselector Tags.
14599 * gst/playback/gstplaybasebin.c: (set_active_source):
14600 Streamselector now selects pads based on the pad object instead of its
14602 * gst/playback/gstplaybin2.c: (gst_play_bin_class_init),
14603 (init_group), (gst_play_bin_init), (get_group), (get_tags),
14604 (gst_play_bin_get_video_tags), (gst_play_bin_get_audio_tags),
14605 (gst_play_bin_get_text_tags),
14606 (gst_play_bin_set_current_video_stream),
14607 (gst_play_bin_set_current_audio_stream),
14608 (gst_play_bin_set_current_text_stream),
14609 (gst_play_bin_set_property), (gst_play_bin_get_property),
14610 (pad_added_cb), (pad_removed_cb), (autoplug_select_cb):
14611 Remove option to mute streams with the current-a/v/t property, we have
14612 this functionality in the flags.
14613 Add signals to notify when the number of A/V/T channels changed.
14614 Add action signals to get tags for the A/V/T streams.
14615 Implement setting the current A/V/T stream.
14616 Rearrange some things to simplify stream selection.
14618 * gst/playback/gstplaysink.c: (gst_play_sink_set_volume),
14619 (gst_play_sink_get_volume), (gst_play_sink_set_property),
14620 (gst_play_sink_get_property), (gen_video_chain), (gen_audio_chain),
14621 (activate_vis), (gst_play_sink_reconfigure):
14622 * gst/playback/gstplaysink.h:
14623 Add and implement volume setting methods.
14624 * gst/playback/gststreamselector.c: (gst_selector_pad_class_init),
14625 (gst_selector_pad_finalize), (gst_selector_pad_get_property),
14626 (gst_selector_pad_event), (gst_stream_selector_class_init),
14627 (gst_stream_selector_init), (gst_stream_selector_finalize),
14628 (gst_stream_selector_set_property),
14629 (gst_stream_selector_get_property),
14630 (gst_stream_selector_get_linked_pad),
14631 (gst_stream_selector_request_new_pad):
14632 * gst/playback/gststreamselector.h:
14633 Add pad properties for tags and status of pads.
14635 Make active pad selection based on pad object instead of name.
14637 2008-02-08 16:10:25 +0000 Stefan Kost <ensonic@users.sourceforge.net>
14639 configure.ac: Revert last change as we now check in gtk-doc.m4 for sed.
14640 Original commit message from CVS:
14642 Revert last change as we now check in gtk-doc.m4 for sed.
14644 2008-02-08 14:54:30 +0000 Jan Schmidt <thaytan@mad.scientist.com>
14646 configure.ac: Find and subst SED when building the docs.
14647 Original commit message from CVS:
14649 Find and subst SED when building the docs.
14651 2008-02-08 14:34:41 +0000 Julien Moutte <julien@moutte.net>
14653 tests/examples/seek/seek.c: Make sure bus signals are reconnected when pressing STOP and then PLAY again for a parse ...
14654 Original commit message from CVS:
14655 2008-02-08 Julien Moutte <julien@fluendo.com>
14656 * tests/examples/seek/seek.c: (stop_cb), (connect_bus_signals),
14657 (main): Make sure bus signals are reconnected when pressing STOP
14658 and then PLAY again for a parse launch pipeline. Fix a ref leak
14660 * win32/common/config.h: Updated.
14662 2008-02-08 00:57:21 +0000 Jan Schmidt <thaytan@mad.scientist.com>
14664 configure.ac: Make DISABLE_DEPRECATED defined *only* during CVS, not during pre-releases or releases.
14665 Original commit message from CVS:
14667 Make DISABLE_DEPRECATED defined *only* during CVS, not during
14668 pre-releases or releases.
14670 2008-02-08 00:45:56 +0000 Jan Schmidt <thaytan@mad.scientist.com>
14672 Subst GIO_LDFLAGS to avoid undefined Makefile var error Zaheer is reporting
14673 Original commit message from CVS:
14675 * ext/gio/Makefile.am:
14676 Subst GIO_LDFLAGS to avoid undefined Makefile var error Zaheer is
14679 2008-02-07 23:40:30 +0000 Jan Schmidt <thaytan@mad.scientist.com>
14681 docs/plugins/Makefile.am: Add the headers which need scanning for the GIO plugin. The rest of the docs still need mig...
14682 Original commit message from CVS:
14683 * docs/plugins/Makefile.am:
14684 Add the headers which need scanning for the GIO plugin. The rest of
14685 the docs still need migrating.
14687 2008-02-07 23:22:23 +0000 Jan Schmidt <thaytan@mad.scientist.com>
14689 Add gio in a few more places.
14690 Original commit message from CVS:
14692 * tests/check/Makefile.am:
14693 * tests/check/pipelines/.cvsignore:
14694 Add gio in a few more places.
14696 2008-02-07 23:18:43 +0000 Jan Schmidt <thaytan@mad.scientist.com>
14698 Move gio plugin from -bad and mark as experimental.
14699 Original commit message from CVS:
14702 * tests/check/Makefile.am:
14703 Move gio plugin from -bad and mark as experimental.
14705 2008-02-07 22:39:00 +0000 Jan Schmidt <thaytan@mad.scientist.com>
14707 gst-libs/gst/interfaces/: Comment out a couple of other things which break the build when
14708 Original commit message from CVS:
14709 * gst-libs/gst/interfaces/mixeroptions.c:
14710 * gst-libs/gst/interfaces/mixertrack.c:
14711 Comment out a couple of other things which break the build when
14712 GST_DISABLE_DEPRECATED isn't on but -Werror is.
14714 2008-02-07 18:28:29 +0000 Tim-Philipp Müller <tim@centricular.net>
14716 docs/libs/gst-plugins-base-libs-sections.txt: Fix pbutils header.
14717 Original commit message from CVS:
14718 * docs/libs/gst-plugins-base-libs-sections.txt:
14719 Fix pbutils header.
14721 2008-02-07 18:07:41 +0000 Christian Schaller <uraeus@gnome.org>
14723 * gst-plugins-base.spec.in:
14724 commit spec file update which includes all the split .pc files
14725 Original commit message from CVS:
14726 commit spec file update which includes all the split .pc files
14728 2008-02-07 12:17:49 +0000 Wim Taymans <wim.taymans@gmail.com>
14730 gst-libs/gst/rtsp/gstrtspmessage.c: Fix compiler warning.
14731 Original commit message from CVS:
14732 * gst-libs/gst/rtsp/gstrtspmessage.c: (gst_rtsp_message_unset):
14733 Fix compiler warning.
14735 2008-02-07 11:00:45 +0000 Peter Kjellerstedt <pkj@axis.com>
14737 gst-libs/gst/sdp/gstsdpmessage.c: Clear the addrinfo struct using memset. Fixes #514937.
14738 Original commit message from CVS:
14739 Patch by: Peter Kjellerstedt <pkj at axis com>
14740 * gst-libs/gst/sdp/gstsdpmessage.c: (is_multicast_address):
14741 Clear the addrinfo struct using memset. Fixes #514937.
14743 2008-02-06 15:07:30 +0000 Wim Taymans <wim.taymans@gmail.com>
14745 gst/tcp/gstfdset.h: Remove unused field to same some memory.
14746 Original commit message from CVS:
14747 * gst/tcp/gstfdset.h:
14748 Remove unused field to same some memory.
14749 * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init):
14750 Mark action signals as such.
14752 2008-02-06 13:35:58 +0000 Michael Smith <msmith@xiph.org>
14754 ext/theora/theoradec.c: Increment granulepos for new-bitstream versions appropriately.
14755 Original commit message from CVS:
14756 * ext/theora/theoradec.c: (_theora_granule_frame),
14758 Increment granulepos for new-bitstream versions appropriately.
14761 2008-02-04 11:51:31 +0000 Wim Taymans <wim.taymans@gmail.com>
14763 tests/examples/seek/seek.c: Remove obsolete stream_time reset after flushing seek, core does that automatically now.
14764 Original commit message from CVS:
14765 * tests/examples/seek/seek.c: (do_seek),
14766 (rate_spinbutton_changed_cb), (update_streams), (main):
14767 Remove obsolete stream_time reset after flushing seek, core does that
14769 Improve accuracy of speed spinbutton.
14770 Only do playbin2 stuff when we actually use it.
14772 2008-02-02 17:29:32 +0000 Tim-Philipp Müller <tim@centricular.net>
14774 tests/check/Makefile.am: Revert previous change of the test environment's GST_PLUGIN_PATH.
14775 Original commit message from CVS:
14776 * tests/check/Makefile.am:
14777 Revert previous change of the test environment's GST_PLUGIN_PATH.
14778 The problem is not with the plugins, but with element factories
14779 and only occurs if elements are split out from existing plugins
14780 or if plugins change name (see #512740).
14782 2008-02-02 15:32:23 +0000 Tim-Philipp Müller <tim@centricular.net>
14784 tests/check/Makefile.am: Fix the tests environment's GST_PLUGIN_PATH: we want the directory with the core's plugins f...
14785 Original commit message from CVS:
14786 * tests/check/Makefile.am:
14787 Fix the tests environment's GST_PLUGIN_PATH: we want the directory
14788 with the core's plugins first and our local build directories last,
14789 since we might be building against an installed core, and that
14790 core's plugin directory may contain older or other versions of
14791 our own -base plugins, but we really do want to test our local
14792 ones (if there are multiple plugins or element factories with the
14793 same name, those inspected last will trump those read in earlier).
14794 Fixes #512740 for the most part.
14796 2008-02-02 07:13:15 +0000 Sebastian Dröge <slomo@circular-chaos.org>
14798 Use gmtime_r if available as gmtime is not MT-safe.
14799 Original commit message from CVS:
14801 * gst-libs/gst/rtsp/gstrtspconnection.c: (add_date_header):
14802 Use gmtime_r if available as gmtime is not MT-safe.
14805 2008-02-02 06:52:41 +0000 Sebastian Dröge <slomo@circular-chaos.org>
14807 gst-libs/gst/rtsp/gstrtspconnection.c: Cast glong to time_t as time_t might have a different type on other platforms,...
14808 Original commit message from CVS:
14809 * gst-libs/gst/rtsp/gstrtspconnection.c: (add_date_header):
14810 Cast glong to time_t as time_t might have a different type on
14811 other platforms, like FreeBSD, and we get a compiler warning
14812 otherwise. Fixes bug #511825.
14814 2008-02-01 16:44:21 +0000 Wim Taymans <wim.taymans@gmail.com>
14816 gst/playback/gstplaybin2.c: Remove stream-info, we going for something easier.
14817 Original commit message from CVS:
14818 * gst/playback/gstplaybin2.c: (gst_play_bin_class_init),
14819 (get_group), (get_n_pads), (gst_play_bin_get_property),
14820 (pad_added_cb), (no_more_pads_cb), (perform_eos),
14821 (autoplug_select_cb), (deactivate_group):
14822 Remove stream-info, we going for something easier.
14823 Refactor getting the current group.
14824 Implement getting the number of audio/video/text streams.
14825 * gst/playback/gststreamselector.c:
14826 (gst_stream_selector_class_init), (gst_stream_selector_init),
14827 (gst_stream_selector_get_property),
14828 (gst_stream_selector_request_new_pad),
14829 (gst_stream_selector_release_pad):
14830 * gst/playback/gststreamselector.h:
14831 Add property for number of pads.
14832 * tests/examples/seek/seek.c: (set_scale), (update_flag),
14833 (vis_toggle_cb), (audio_toggle_cb), (video_toggle_cb),
14834 (text_toggle_cb), (update_streams), (msg_async_done),
14835 (msg_state_changed), (main):
14836 Block slider callback when updating the slider position.
14837 Add gui elements for controlling playbin2.
14838 Add callback for async_done that updates position/duration.
14840 2008-02-01 12:56:59 +0000 Stefan Kost <ensonic@users.sourceforge.net>
14842 docs/plugins/: First round of plugin docs cleansups.
14843 Original commit message from CVS:
14844 * docs/plugins/Makefile.am:
14845 * docs/plugins/gst-plugins-base-plugins-docs.sgml:
14846 * docs/plugins/gst-plugins-base-plugins-sections.txt:
14847 * docs/plugins/gst-plugins-base-plugins.hierarchy:
14848 * docs/plugins/gst-plugins-base-plugins.interfaces:
14849 * docs/plugins/gst-plugins-base-plugins.prerequisites:
14850 First round of plugin docs cleansups.
14851 * docs/plugins/inspect/plugin-adder.xml:
14852 * docs/plugins/inspect/plugin-alsa.xml:
14853 * docs/plugins/inspect/plugin-audioconvert.xml:
14854 * docs/plugins/inspect/plugin-audiorate.xml:
14855 * docs/plugins/inspect/plugin-audioresample.xml:
14856 * docs/plugins/inspect/plugin-audiotestsrc.xml:
14857 * docs/plugins/inspect/plugin-cdparanoia.xml:
14858 * docs/plugins/inspect/plugin-decodebin.xml:
14859 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
14860 * docs/plugins/inspect/plugin-gdp.xml:
14861 * docs/plugins/inspect/plugin-gnomevfs.xml:
14862 * docs/plugins/inspect/plugin-libvisual.xml:
14863 * docs/plugins/inspect/plugin-ogg.xml:
14864 * docs/plugins/inspect/plugin-pango.xml:
14865 * docs/plugins/inspect/plugin-subparse.xml:
14866 * docs/plugins/inspect/plugin-tcp.xml:
14867 * docs/plugins/inspect/plugin-theora.xml:
14868 * docs/plugins/inspect/plugin-typefindfunctions.xml:
14869 * docs/plugins/inspect/plugin-video4linux.xml:
14870 * docs/plugins/inspect/plugin-videorate.xml:
14871 * docs/plugins/inspect/plugin-videoscale.xml:
14872 * docs/plugins/inspect/plugin-videotestsrc.xml:
14873 * docs/plugins/inspect/plugin-volume.xml:
14874 * docs/plugins/inspect/plugin-vorbis.xml:
14875 * docs/plugins/inspect/plugin-ximagesink.xml:
14876 * docs/plugins/inspect/plugin-xvimagesink.xml:
14878 * ext/ogg/Makefile.am:
14879 * ext/ogg/gstoggmux.c:
14880 * ext/ogg/gstoggmux.h:
14881 Add header for oggmux. the c-file needs a doc blob still.
14883 2008-02-01 11:09:16 +0000 Thijs Vermeir <thijsvermeir@gmail.com>
14885 Add gst_rtp_buffer_set_extension_data()
14886 Original commit message from CVS:
14887 Patch by: Thijs Vermeir <thijsvermeir at gmail dot com>
14888 * gst-libs/gst/rtp/gstrtpbuffer.c:
14889 (gst_rtp_buffer_set_extension_data):
14890 * gst-libs/gst/rtp/gstrtpbuffer.h:
14891 * tests/check/libs/rtp.c: (GST_START_TEST), (rtp_suite):
14892 Add gst_rtp_buffer_set_extension_data()
14893 Add a unit test for this addition. Fixes #511478.
14894 API: GstRTPBuffer:gst_rtp_buffer_set_extension_data()
14896 2008-01-31 17:18:46 +0000 Wim Taymans <wim.taymans@gmail.com>
14898 gst-libs/gst/app/gstappsink.c: Really clean up the queue instead of just unreffing all buffers in it.
14899 Original commit message from CVS:
14900 * gst-libs/gst/app/gstappsink.c: (gst_app_sink_dispose):
14901 Really clean up the queue instead of just unreffing all buffers
14903 * gst-libs/gst/app/gstappsrc.c: (gst_app_src_base_init),
14904 (gst_app_src_class_init), (gst_app_src_init),
14905 (gst_app_src_dispose), (gst_app_src_finalize):
14906 Fix dispose/finalize.
14908 2008-01-30 15:34:25 +0000 Sebastian Dröge <slomo@circular-chaos.org>
14910 ext/gio/: Use async variants of the close stream functions to prevent blocking for a long time there and add some mor...
14911 Original commit message from CVS:
14912 * ext/gio/gstgiobasesink.c: (close_stream_cb),
14913 (gst_gio_base_sink_stop), (gst_gio_base_sink_event),
14914 (gst_gio_base_sink_render), (gst_gio_base_sink_set_stream):
14915 * ext/gio/gstgiobasesrc.c: (close_stream_cb),
14916 (gst_gio_base_src_stop), (gst_gio_base_src_create),
14917 (gst_gio_base_src_set_stream):
14918 Use async variants of the close stream functions to prevent blocking
14919 for a long time there and add some more sanity checks for a correct
14922 2008-01-30 14:42:14 +0000 Jan Schmidt <thaytan@mad.scientist.com>
14924 configure.ac: Back to CVS
14925 Original commit message from CVS:
14929 === release 0.10.17 ===
14931 2008-01-30 14:19:05 +0000 Jan Schmidt <thaytan@mad.scientist.com>
14937 * docs/plugins/gst-plugins-base-plugins.hierarchy:
14938 * docs/plugins/inspect/plugin-adder.xml:
14939 * docs/plugins/inspect/plugin-alsa.xml:
14940 * docs/plugins/inspect/plugin-audioconvert.xml:
14941 * docs/plugins/inspect/plugin-audiorate.xml:
14942 * docs/plugins/inspect/plugin-audioresample.xml:
14943 * docs/plugins/inspect/plugin-audiotestsrc.xml:
14944 * docs/plugins/inspect/plugin-cdparanoia.xml:
14945 * docs/plugins/inspect/plugin-decodebin.xml:
14946 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
14947 * docs/plugins/inspect/plugin-gdp.xml:
14948 * docs/plugins/inspect/plugin-gnomevfs.xml:
14949 * docs/plugins/inspect/plugin-libvisual.xml:
14950 * docs/plugins/inspect/plugin-ogg.xml:
14951 * docs/plugins/inspect/plugin-pango.xml:
14952 * docs/plugins/inspect/plugin-subparse.xml:
14953 * docs/plugins/inspect/plugin-tcp.xml:
14954 * docs/plugins/inspect/plugin-theora.xml:
14955 * docs/plugins/inspect/plugin-typefindfunctions.xml:
14956 * docs/plugins/inspect/plugin-video4linux.xml:
14957 * docs/plugins/inspect/plugin-videorate.xml:
14958 * docs/plugins/inspect/plugin-videoscale.xml:
14959 * docs/plugins/inspect/plugin-videotestsrc.xml:
14960 * docs/plugins/inspect/plugin-volume.xml:
14961 * docs/plugins/inspect/plugin-vorbis.xml:
14962 * docs/plugins/inspect/plugin-ximagesink.xml:
14963 * docs/plugins/inspect/plugin-xvimagesink.xml:
14964 * gst-plugins-base.doap:
14965 * win32/common/config.h:
14967 Original commit message from CVS:
14970 2008-01-30 13:45:27 +0000 Jan Schmidt <thaytan@mad.scientist.com>
14972 gst-libs/gst/interfaces/: Also remove the conditional registration of the signals that disappeared with the ABI chang...
14973 Original commit message from CVS:
14974 * gst-libs/gst/interfaces/mixeroptions.c:
14975 * gst-libs/gst/interfaces/mixertrack.c:
14976 Also remove the conditional registration of the signals
14977 that disappeared with the ABI change in 0.10.14
14979 2008-01-30 12:28:59 +0000 Jan Schmidt <thaytan@mad.scientist.com>
14981 gst-libs/gst/rtsp/gstrtspconnection.c: Revert patch to gstrtspconnection.c for brown paper bag release of -base. Re-o...
14982 Original commit message from CVS:
14983 * gst-libs/gst/rtsp/gstrtspconnection.c:
14984 Revert patch to gstrtspconnection.c for brown paper bag
14985 release of -base. Re-opens: #511825
14987 2008-01-30 12:20:42 +0000 Jan Schmidt <thaytan@mad.scientist.com>
14989 gst-libs/gst/interfaces/: Change the way these deprecated function pointers are removed so that the compiled ABI is u...
14990 Original commit message from CVS:
14991 * gst-libs/gst/interfaces/mixeroptions.h:
14992 * gst-libs/gst/interfaces/mixertrack.h:
14993 Change the way these deprecated function pointers are removed
14994 so that the compiled ABI is unconditionally smaller. This
14995 sets in stone an ABI break that actually occurred when the
14996 things were deprecated in 0.10.14, which seems to be the best
14997 fix as the only known users are oss-mixer and sunaudio-mixer in
15001 2008-01-30 12:19:02 +0000 Jan Schmidt <thaytan@mad.scientist.com>
15003 gst-libs/gst/interfaces/: Change the way these deprecated function pointers are removed so that the compiled ABI is u...
15004 Original commit message from CVS:
15005 * gst-libs/gst/interfaces/mixeroptions.h:
15006 * gst-libs/gst/interfaces/mixertrack.h:
15007 Change the way these deprecated function pointers are removed
15008 so that the compiled ABI is unconditionally smaller. This
15009 sets in stone an ABI break that actually occurred when the
15010 things were deprecated in 0.10.14, which seems to be the best
15011 fix as the only known users are oss-mixer and sunaudio-mixer in
15014 2008-01-30 11:43:53 +0000 Tim-Philipp Müller <tim@centricular.net>
15016 win32/common/libgstpbutils.def: Export the two new _get_type() functions which are needed by the python bindings.
15017 Original commit message from CVS:
15018 * win32/common/libgstpbutils.def:
15019 Export the two new _get_type() functions which are needed
15020 by the python bindings.
15022 2008-01-29 09:59:03 +0000 Sebastian Dröge <slomo@circular-chaos.org>
15024 gst-libs/gst/rtsp/gstrtspconnection.c: Cast glong to time_t as time_t might have a different type on other platforms,...
15025 Original commit message from CVS:
15026 * gst-libs/gst/rtsp/gstrtspconnection.c: (add_date_header):
15027 Cast glong to time_t as time_t might have a different type on
15028 other platforms, like FreeBSD, and we get a compiler warning
15029 otherwise. Fixes bug #511825.
15031 2008-01-29 09:47:12 +0000 Sebastian Dröge <slomo@circular-chaos.org>
15033 gst-libs/gst/audio/gstaudiofilter.c: Initialize the GstRingerBuffer class to get it's debug category initialized. gst...
15034 Original commit message from CVS:
15035 * gst-libs/gst/audio/gstaudiofilter.c:
15036 (gst_audio_filter_class_init):
15037 Initialize the GstRingerBuffer class to get it's debug category
15038 initialized. gst_ring_buffer_parse_caps() uses the ringbuffer debug
15039 category and otherwise we get some g_critical(). Fixes bug #512334.
15041 2008-01-28 23:35:21 +0000 Jan Schmidt <thaytan@mad.scientist.com>
15043 configure.ac: Back to CVS
15044 Original commit message from CVS:
15048 === release 0.10.16 ===
15050 2008-01-28 23:31:26 +0000 Jan Schmidt <thaytan@mad.scientist.com>
15056 * docs/plugins/gst-plugins-base-plugins.args:
15057 * docs/plugins/gst-plugins-base-plugins.hierarchy:
15058 * docs/plugins/gst-plugins-base-plugins.interfaces:
15059 * docs/plugins/gst-plugins-base-plugins.prerequisites:
15060 * docs/plugins/gst-plugins-base-plugins.signals:
15061 * docs/plugins/inspect/plugin-adder.xml:
15062 * docs/plugins/inspect/plugin-alsa.xml:
15063 * docs/plugins/inspect/plugin-audioconvert.xml:
15064 * docs/plugins/inspect/plugin-audiorate.xml:
15065 * docs/plugins/inspect/plugin-audioresample.xml:
15066 * docs/plugins/inspect/plugin-audiotestsrc.xml:
15067 * docs/plugins/inspect/plugin-cdparanoia.xml:
15068 * docs/plugins/inspect/plugin-decodebin.xml:
15069 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
15070 * docs/plugins/inspect/plugin-gdp.xml:
15071 * docs/plugins/inspect/plugin-gnomevfs.xml:
15072 * docs/plugins/inspect/plugin-libvisual.xml:
15073 * docs/plugins/inspect/plugin-ogg.xml:
15074 * docs/plugins/inspect/plugin-pango.xml:
15075 * docs/plugins/inspect/plugin-subparse.xml:
15076 * docs/plugins/inspect/plugin-tcp.xml:
15077 * docs/plugins/inspect/plugin-theora.xml:
15078 * docs/plugins/inspect/plugin-typefindfunctions.xml:
15079 * docs/plugins/inspect/plugin-video4linux.xml:
15080 * docs/plugins/inspect/plugin-videorate.xml:
15081 * docs/plugins/inspect/plugin-videoscale.xml:
15082 * docs/plugins/inspect/plugin-videotestsrc.xml:
15083 * docs/plugins/inspect/plugin-volume.xml:
15084 * docs/plugins/inspect/plugin-vorbis.xml:
15085 * docs/plugins/inspect/plugin-ximagesink.xml:
15086 * docs/plugins/inspect/plugin-xvimagesink.xml:
15087 * gst-plugins-base.doap:
15088 * win32/common/config.h:
15090 Original commit message from CVS:
15093 2008-01-28 22:15:47 +0000 Jan Schmidt <thaytan@mad.scientist.com>
15119 Original commit message from CVS:
15122 2008-01-22 15:37:49 +0000 Thijs Vermeir <thijsvermeir@gmail.com>
15124 gst-libs/gst/rtp/gstrtpbuffer.c: Fix typos and wrong extension check. Fixes #511274.
15125 Original commit message from CVS:
15126 Patch by: Thijs Vermeir <thijsvermeir at gmail dot com>
15127 * gst-libs/gst/rtp/gstrtpbuffer.c:
15128 (gst_rtp_buffer_get_extension_data):
15129 Fix typos and wrong extension check. Fixes #511274.
15131 2008-01-18 00:03:18 +0000 Jan Schmidt <thaytan@mad.scientist.com>
15133 po/sk.po: Oops - add new sk.po mentioned in the LINGUAS I just committed
15134 Original commit message from CVS:
15136 Oops - add new sk.po mentioned in the LINGUAS I just committed
15138 2008-01-17 22:31:25 +0000 Jan Schmidt <thaytan@mad.scientist.com>
15140 po/LINGUAS: Add ca translation to the disted list.
15141 Original commit message from CVS:
15143 Add ca translation to the disted list.
15144 * win32/vs6/libgstsdp.dsp:
15145 Convert line endings to CRLF
15147 2008-01-17 21:58:53 +0000 Sébastien Moutte <sebastien@moutte.net>
15149 win32/MANIFEST: Add win32/vs6/libgstrtsp.dsp to MANIFEST
15150 Original commit message from CVS:
15152 Add win32/vs6/libgstrtsp.dsp to MANIFEST
15154 2008-01-16 05:40:48 +0000 Sebastian Dröge <slomo@circular-chaos.org>
15156 Update for API changes in GIO and require GIO 2.15.2 for this.
15157 Original commit message from CVS:
15159 * tests/check/pipelines/gio.c: (GST_START_TEST):
15160 Update for API changes in GIO and require GIO 2.15.2 for this.
15162 2008-01-14 22:20:12 +0000 Jan Schmidt <thaytan@mad.scientist.com>
15164 win32/common/: Add new API declarations
15165 Original commit message from CVS:
15166 * win32/common/libgstsdp.def:
15167 * win32/common/libgstvideo.def:
15168 Add new API declarations
15170 2008-01-14 17:00:03 +0000 Jan Schmidt <thaytan@mad.scientist.com>
15172 ext/theora/: Take a 2nd stab at handling libtheora granulepos changes in the decoder and parser by inspecting the bit...
15173 Original commit message from CVS:
15174 * ext/theora/gsttheoradec.h:
15175 * ext/theora/gsttheoraparse.h:
15176 * ext/theora/theoradec.c:
15177 * ext/theora/theoraparse.c:
15178 Take a 2nd stab at handling libtheora granulepos changes in the decoder
15179 and parser by inspecting the bitstream version of the incoming data.
15181 2008-01-14 13:11:05 +0000 Sebastian Dröge <slomo@circular-chaos.org>
15183 Provide one pkg-config file for every gst-plugins-base library.
15184 Original commit message from CVS:
15186 * pkgconfig/Makefile.am:
15187 * pkgconfig/gstreamer-audio-uninstalled.pc.in:
15188 * pkgconfig/gstreamer-audio.pc.in:
15189 * pkgconfig/gstreamer-cdda-uninstalled.pc.in:
15190 * pkgconfig/gstreamer-cdda.pc.in:
15191 * pkgconfig/gstreamer-fft-uninstalled.pc.in:
15192 * pkgconfig/gstreamer-fft.pc.in:
15193 * pkgconfig/gstreamer-floatcast-uninstalled.pc.in:
15194 * pkgconfig/gstreamer-floatcast.pc.in:
15195 * pkgconfig/gstreamer-interfaces-uninstalled.pc.in:
15196 * pkgconfig/gstreamer-interfaces.pc.in:
15197 * pkgconfig/gstreamer-netbuffer-uninstalled.pc.in:
15198 * pkgconfig/gstreamer-netbuffer.pc.in:
15199 * pkgconfig/gstreamer-pbutils-uninstalled.pc.in:
15200 * pkgconfig/gstreamer-pbutils.pc.in:
15201 * pkgconfig/gstreamer-riff-uninstalled.pc.in:
15202 * pkgconfig/gstreamer-riff.pc.in:
15203 * pkgconfig/gstreamer-rtp-uninstalled.pc.in:
15204 * pkgconfig/gstreamer-rtp.pc.in:
15205 * pkgconfig/gstreamer-rtsp-uninstalled.pc.in:
15206 * pkgconfig/gstreamer-rtsp.pc.in:
15207 * pkgconfig/gstreamer-sdp-uninstalled.pc.in:
15208 * pkgconfig/gstreamer-sdp.pc.in:
15209 * pkgconfig/gstreamer-tag-uninstalled.pc.in:
15210 * pkgconfig/gstreamer-tag.pc.in:
15211 * pkgconfig/gstreamer-video-uninstalled.pc.in:
15212 * pkgconfig/gstreamer-video.pc.in:
15213 Provide one pkg-config file for every gst-plugins-base library.
15214 This makes linking to those libraries much more intuitive and
15215 provides standard pkg-config behaviour for them. Fixes bug #499697.
15217 2008-01-14 01:19:34 +0000 David Schleef <ds@schleef.org>
15219 gst/videoscale/vs_4tap.c: Fix valgrind error on 4tap scaling method.
15220 Original commit message from CVS:
15221 * gst/videoscale/vs_4tap.c:
15222 Fix valgrind error on 4tap scaling method.
15224 2008-01-13 21:40:45 +0000 Sébastien Moutte <sebastien@moutte.net>
15226 gst-libs/gst/sdp/gstsdpmessage.c: Include Winsock2.h for VS6 and use a different way initialize hints structure so it...
15227 Original commit message from CVS:
15228 * gst-libs/gst/sdp/gstsdpmessage.c: (is_multicast_address):
15229 Include Winsock2.h for VS6 and use a different way initialize
15230 hints structure so it can build with VS6.
15232 * win32/vs6/libgstsdp.dsp:
15233 * win32/common/libgstsdp.def:
15234 Add new files for libgstsdp.
15235 * win32/vs6/grammar.dsp:
15236 Copy pbutils-enumtypes* from win32/common to pbutils sources folder.
15237 * win32/vs6/gst_plugins_base.dsw:
15238 * win32/vs6/libgstdecodebin.dsp:
15239 * win32/vs6/libgstdecodebin2.dsp:
15240 * win32/vs6/libgstplaybin.dsp:
15241 * win32/vs6/libgstvolume.dsp:
15242 Add new dependencies to the link list.
15244 2008-01-13 17:24:49 +0000 Julien Moutte <julien@moutte.net>
15246 win32/common/: Update/Add generated files in the win32 build directory.
15247 Original commit message from CVS:
15248 2008-01-13 Julien Moutte <julien@fluendo.com>
15249 * win32/common/config.h:
15250 * win32/common/gstrtsp-enumtypes.c: (gst_rtsp_result_get_type),
15251 (gst_rtsp_event_get_type), (gst_rtsp_family_get_type),
15252 (gst_rtsp_state_get_type), (gst_rtsp_version_get_type),
15253 (gst_rtsp_method_get_type), (gst_rtsp_auth_method_get_type),
15254 (gst_rtsp_header_field_get_type),
15255 (gst_rtsp_status_code_get_type):
15256 * win32/common/interfaces-enumtypes.c:
15257 (gst_color_balance_type_get_type), (gst_mixer_type_get_type),
15258 (gst_mixer_message_type_get_type), (gst_mixer_flags_get_type),
15259 (gst_mixer_track_flags_get_type),
15260 (gst_tuner_channel_flags_get_type):
15261 * win32/common/multichannel-enumtypes.c:
15262 (gst_audio_channel_position_get_type):
15263 * win32/common/pbutils-enumtypes.c:
15264 (gst_install_plugins_return_get_type):
15265 * win32/common/pbutils-enumtypes.h: Update/Add generated files
15266 in the win32 build directory.
15268 2008-01-12 23:24:02 +0000 Jan Schmidt <thaytan@mad.scientist.com>
15270 tests/check/Makefile.am: Fix CFLAGS to also pull in the gstcheck cflags from AM_CFLAGS.
15271 Original commit message from CVS:
15272 * tests/check/Makefile.am:
15273 Fix CFLAGS to also pull in the gstcheck cflags from AM_CFLAGS.
15274 * tests/check/elements/audiorate.c: (do_perfect_stream_test):
15275 * tests/check/elements/playbin.c:
15276 * tests/check/libs/mixer.c: (test_element_interface_supported),
15277 (gst_implements_interface_init):
15278 * tests/check/libs/rtp.c: (GST_START_TEST):
15279 Fix various assignment type mismatches.
15281 2008-01-12 23:08:28 +0000 Jan Schmidt <thaytan@mad.scientist.com>
15283 Add test to see if hstrerror is available or if we need libresolv (Solaris) for it, then use it in libgstrtsp.
15284 Original commit message from CVS:
15286 * gst-libs/gst/rtsp/Makefile.am:
15287 Add test to see if hstrerror is available or if we need libresolv
15288 (Solaris) for it, then use it in libgstrtsp.
15290 2008-01-12 14:54:51 +0000 Jan Schmidt <thaytan@mad.scientist.com>
15292 gst-libs/gst/tag/Makefile.am: Fix include path order
15293 Original commit message from CVS:
15294 * gst-libs/gst/tag/Makefile.am:
15295 Fix include path order
15297 2008-01-11 17:15:23 +0000 Tim-Philipp Müller <tim@centricular.net>
15299 * gst-libs/gst/pbutils/.gitignore:
15300 Ignore more and make buildbot happy
15301 Original commit message from CVS:
15302 Ignore more and make buildbot happy
15304 2008-01-11 16:18:10 +0000 Edward Hervey <bilboed@bilboed.com>
15306 gst-libs/gst/pbutils/install-plugins.*: Add GBoxed GType for GstInstallPluginsContext, this eases the wrapping for bi...
15307 Original commit message from CVS:
15308 * gst-libs/gst/pbutils/install-plugins.c:
15309 (gst_install_plugins_context_copy),
15310 (gst_install_plugins_context_get_type):
15311 * gst-libs/gst/pbutils/install-plugins.h:
15312 Add GBoxed GType for GstInstallPluginsContext, this eases the wrapping
15315 2008-01-11 15:48:11 +0000 Michael Smith <msmith@xiph.org>
15317 ext/theora/theoradec.c: Adapt for post-alpha meaning of granulepos, when we have a newer version of libtheora.
15318 Original commit message from CVS:
15319 * ext/theora/theoradec.c: (gst_theora_dec_class_init),
15320 (_theora_granule_frame), (_theora_granule_start_time),
15321 (theora_dec_sink_convert), (theora_dec_decode_buffer):
15322 Adapt for post-alpha meaning of granulepos, when we
15323 have a newer version of libtheora.
15324 * ext/theora/theoraenc.c: (gst_theora_enc_class_init),
15325 (theora_enc_get_ogg_packet_end_time), (theora_enc_sink_event),
15326 (theora_enc_is_discontinuous), (theora_enc_chain):
15328 * tests/check/Makefile.am:
15329 Link libtheora into theoraenc test so we can check which version of
15330 libtheora we're testing against.
15331 * tests/check/pipelines/theoraenc.c: (check_libtheora),
15332 (check_buffer_granulepos),
15333 (check_buffer_granulepos_from_starttime), (GST_START_TEST),
15335 Adapt tests to check the values that are now defined for theora; make
15336 the tests backwards-adapt the passed values if we're running against an
15340 2008-01-10 17:55:53 +0000 Tim-Philipp Müller <tim@centricular.net>
15342 gst-libs/gst/audio/: Ref audio clock class from a thread-safe context to make sure however unlikely that may be in pr...
15343 Original commit message from CVS:
15344 * gst-libs/gst/audio/gstbaseaudiosink.c:
15345 (gst_base_audio_sink_class_init):
15346 * gst-libs/gst/audio/gstbaseaudiosrc.c:
15347 (gst_base_audio_src_class_init):
15348 Ref audio clock class from a thread-safe context to make sure
15349 we're not bit by GObjects lack of thread-safety here (#349410),
15350 however unlikely that may be in practice.
15352 2008-01-10 12:22:46 +0000 Sebastian Dröge <slomo@circular-chaos.org>
15354 autogen.sh: Add -Wno-portability to the automake parameters to stop warnings about GNU make extensions being used. We...
15355 Original commit message from CVS:
15357 Add -Wno-portability to the automake parameters to stop warnings
15358 about GNU make extensions being used. We require GNU make in almost
15359 every Makefile anyway.
15361 Use AM_PROG_CC_C_O as a compiler that accepts both -c and -o
15362 at the same time is required for per target flags.
15364 2008-01-08 21:10:02 +0000 Tim-Philipp Müller <tim@centricular.net>
15366 gst-libs/gst/tag/gsttagdemux.c: Post an error message if we can't pull as many bytes as we need for the tag. This mak...
15367 Original commit message from CVS:
15368 * gst-libs/gst/tag/gsttagdemux.c: (gst_tag_demux_pull_start_tag):
15369 Post an error message if we can't pull as many bytes as we need
15370 for the tag. This makes sure the user gets to see a proper error
15371 message if a file with a partial ID3 tag is fed to decodebin, and
15372 not a 'no ID3 tag demuxer' error, which would be confusing
15375 2008-01-08 20:59:20 +0000 Tim-Philipp Müller <tim@centricular.net>
15377 gst-libs/gst/pbutils/descriptions.c: Add description strings for ID3, APE, and ICY tags.
15378 Original commit message from CVS:
15379 * gst-libs/gst/pbutils/descriptions.c: (formats):
15380 Add description strings for ID3, APE, and ICY tags.
15382 2008-01-08 20:48:00 +0000 Tim-Philipp Müller <tim@centricular.net>
15384 gst/playback/gstdecodebin.c: Make sure we error out correctly if we can't activate one of the elements we've added. ...
15385 Original commit message from CVS:
15386 * gst/playback/gstdecodebin.c: (try_to_link_1):
15387 Make sure we error out correctly if we can't activate one of
15388 the elements we've added. Fixes #508138.
15390 2008-01-07 13:59:43 +0000 Bastien Nocera <hadess@hadess.net>
15392 ext/alsa/gstalsamixer.c: Use snd_mixer_selem_set_{playback|capture}_volume_all() if the volume is the same for all ch...
15393 Original commit message from CVS:
15394 Patch by: Bastien Nocera <hadess at hadess net>
15395 * ext/alsa/gstalsamixer.c: (gst_alsa_mixer_get_volume),
15396 (check_if_volumes_are_the_same), (gst_alsa_mixer_set_volume):
15397 Use snd_mixer_selem_set_{playback|capture}_volume_all() if
15398 the volume is the same for all channels. This works around
15399 some problem in alsa that leaves us with inconsistent state
15400 for some reason (#486840).
15402 2008-01-07 13:19:50 +0000 Jerone Young <jerone@gmail.com>
15404 ext/alsa/gstalsamixer.c: If there's no mixer track by the name of 'Master' or 'Front', check if there's one called 'P...
15405 Original commit message from CVS:
15406 Patch by: Jerone Young <jerone at gmail com>
15407 * ext/alsa/gstalsamixer.c: (gst_alsa_mixer_find_master_mixer):
15408 If there's no mixer track by the name of 'Master' or 'Front',
15409 check if there's one called 'PCM' before trying the generic
15410 fallback logic (fixes #506928, where we pick 'Mic' as master
15411 track for the AD1984 card in a Thinkpad T61/X61 laptop).
15413 2008-01-07 11:40:04 +0000 Wim Taymans <wim.taymans@gmail.com>
15415 gst/playback/gstplay-enum.*: Add enums for configuration flags.
15416 Original commit message from CVS:
15417 * gst/playback/gstplay-enum.c:
15418 (register_gst_autoplug_select_result),
15419 (gst_autoplug_select_result_get_type), (register_gst_play_flags),
15420 (gst_play_flags_get_type):
15421 * gst/playback/gstplay-enum.h:
15422 Add enums for configuration flags.
15423 * gst/playback/gstplaybin2.c: (gst_play_bin_class_init),
15424 (init_group), (gst_play_bin_init), (gst_play_bin_set_property),
15425 (gst_play_bin_get_property), (no_more_pads_cb),
15426 (autoplug_select_cb), (gst_play_bin_change_state):
15427 Merge mode with flags.
15428 Add more property getters/setters, defaults and docs.
15429 Add properties to get number of audio/video/text streams.
15430 Create sink object in _init so that we can always rely on it being
15432 * gst/playback/gstplaysink.c: (gst_play_sink_init),
15433 (gen_video_chain), (gen_audio_chain), (gen_vis_chain),
15434 (activate_vis), (gst_play_sink_reconfigure),
15435 (gst_play_sink_set_flags), (gst_play_sink_get_flags),
15436 (gst_play_sink_change_state):
15437 * gst/playback/gstplaysink.h:
15438 Use flags to configure the sink pipelines.
15439 Add tee before audio pipeline so that we can use it for visualisations.
15440 Start working on integrating visualisations.
15441 Remove mode, we can do everything with the flags now.
15442 Add method to configue the sink pipeline.
15444 2008-01-06 16:36:32 +0000 Sebastian Dröge <slomo@circular-chaos.org>
15446 Update to GMemoryInputStream API changes in GLib SVN and require gio-2.0 >= 2.15.1 for this. Fixes bug #507584.
15447 Original commit message from CVS:
15449 * ext/gio/gstgiobasesrc.c: (gst_gio_base_src_get_size):
15450 * tests/check/pipelines/gio.c: (free_input), (GST_START_TEST):
15451 Update to GMemoryInputStream API changes in GLib SVN and require
15452 gio-2.0 >= 2.15.1 for this. Fixes bug #507584.
15453 We can also report the duration for every GSeekable, not only
15454 GFileInputStream and GMemoryInputStream.
15456 2008-01-06 14:39:19 +0000 Tim-Philipp Müller <tim@centricular.net>
15458 tests/check/pipelines/theoraenc.c: Turn these functions into macros so we can see right away where the failure occured.
15459 Original commit message from CVS:
15460 * tests/check/pipelines/theoraenc.c: (check_buffer_is_header),
15461 (check_buffer_timestamp), (check_buffer_duration):
15462 Turn these functions into macros so we can see right away
15463 where the failure occured.
15465 2008-01-05 22:25:05 +0000 Julien Moutte <julien@moutte.net>
15467 sys/xvimage/xvimagesink.c: Add debugging information to understand how X calculates the stride for XvImages.
15468 Original commit message from CVS:
15469 2008-01-05 Julien Moutte <julien@fluendo.com>
15470 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_new): Add
15471 debugging information to understand how X calculates the stride
15474 2008-01-03 20:33:58 +0000 Sebastian Dröge <slomo@circular-chaos.org>
15476 gst/volume/: Use GstAudioFilter as base class for the volume element instead of plain GstBaseTransform.
15477 Original commit message from CVS:
15478 * gst/volume/Makefile.am:
15479 * gst/volume/gstvolume.c: (volume_choose_func),
15480 (gst_volume_base_init), (gst_volume_class_init), (gst_volume_init),
15482 * gst/volume/gstvolume.h:
15483 Use GstAudioFilter as base class for the volume element instead of
15484 plain GstBaseTransform.
15486 2008-01-03 07:17:05 +0000 Sebastian Dröge <slomo@circular-chaos.org>
15488 gst-libs/gst/audio/gstaudiofilter.c: Don't set element details for the abstract GstAudioFilter class.
15489 Original commit message from CVS:
15490 * gst-libs/gst/audio/gstaudiofilter.c: (gst_audio_filter_get_type):
15491 Don't set element details for the abstract GstAudioFilter class.
15493 2008-01-02 12:09:48 +0000 Sebastian Dröge <slomo@circular-chaos.org>
15495 gst-libs/gst/audio/gstaudiofilter.c: Implement get_unit_size() vmethod of GstBaseTransform.
15496 Original commit message from CVS:
15497 * gst-libs/gst/audio/gstaudiofilter.c:
15498 (gst_audio_filter_class_init), (gst_audio_filter_get_unit_size):
15499 Implement get_unit_size() vmethod of GstBaseTransform.
15501 2008-01-01 12:53:48 +0000 Edward Hervey <bilboed@bilboed.com>
15503 gst-libs/gst/pbutils/: Use glib-enum generator to have a proper enum GType for
15504 Original commit message from CVS:
15505 * gst-libs/gst/pbutils/Makefile.am:
15506 * gst-libs/gst/pbutils/pbutils.h:
15507 Use glib-enum generator to have a proper enum GType for
15508 GST_TYPE_INSTALL_PLUGINS_RETURN so we can easily wrap it in bindings.
15510 2008-01-01 01:21:47 +0000 David Schleef <ds@schleef.org>
15512 tests/check/: Reenable theoraenc test, which fails on the buildbot but not locally.
15513 Original commit message from CVS:
15514 * tests/check/Makefile.am:
15515 * tests/check/pipelines/theoraenc.c:
15516 Reenable theoraenc test, which fails on the buildbot but
15519 2007-12-31 21:31:01 +0000 David Schleef <ds@schleef.org>
15521 docs/: Add *-undeclared.txt to fix buildbot.
15522 Original commit message from CVS:
15523 * docs/libs/.cvsignore:
15524 * docs/plugins/.cvsignore:
15525 Add *-undeclared.txt to fix buildbot.
15527 2007-12-31 20:45:28 +0000 David Schleef <ds@schleef.org>
15529 tests/check/Makefile.am: Second attempt at disabling theoraenc test long enough to get buildbot to compile -base.
15530 Original commit message from CVS:
15531 * tests/check/Makefile.am:
15532 Second attempt at disabling theoraenc test long enough to
15533 get buildbot to compile -base.
15535 2007-12-31 20:21:20 +0000 David Schleef <ds@schleef.org>
15537 tests/check/pipelines/theoraenc.c: Disable theoraenc test long enough to get the buildbot to compile a recent -base.
15538 Original commit message from CVS:
15539 * tests/check/pipelines/theoraenc.c:
15540 Disable theoraenc test long enough to get the buildbot to
15541 compile a recent -base.
15543 2007-12-31 13:17:29 +0000 Wim Taymans <wim.taymans@gmail.com>
15545 tests/examples/seek/seek.c: Make sure we reset the slider value to 0.0 without racing against a possible g_idle that ...
15546 Original commit message from CVS:
15547 * tests/examples/seek/seek.c: (stop_cb):
15548 Make sure we reset the slider value to 0.0 without racing against a
15549 possible g_idle that sets it to something else.
15551 2007-12-31 00:32:53 +0000 Thijs Vermeir <thijsvermeir@gmail.com>
15553 sys/ximage/ximagesink.c: fix typo
15554 Original commit message from CVS:
15555 * sys/ximage/ximagesink.c:
15558 2007-12-30 19:21:16 +0000 Wim Taymans <wim.taymans@gmail.com>
15560 gst-libs/gst/rtsp/gstrtspdefs.*: Add Location header so that we can start implementing redirects.
15561 Original commit message from CVS:
15562 * gst-libs/gst/rtsp/gstrtspdefs.c: (rtsp_init_status):
15563 * gst-libs/gst/rtsp/gstrtspdefs.h:
15564 Add Location header so that we can start implementing redirects.
15567 2007-12-29 20:55:39 +0000 Thijs Vermeir <thijsvermeir@gmail.com>
15569 gst/subparse/gstssaparse.c: combine if's
15570 Original commit message from CVS:
15571 * gst/subparse/gstssaparse.c:
15574 2007-12-29 19:23:59 +0000 Thijs Vermeir <thijsvermeir@gmail.com>
15576 gst/subparse/gstssaparse.c: remove duplicate log message
15577 Original commit message from CVS:
15578 * gst/subparse/gstssaparse.c:
15579 remove duplicate log message
15581 2007-12-29 17:29:17 +0000 Sebastian Dröge <slomo@circular-chaos.org>
15583 Update to latest API changes in GLib/GIO and require at least gio-2.0 2.15.0 for this.
15584 Original commit message from CVS:
15586 * ext/gio/gstgio.c:
15587 * ext/gio/gstgio.h:
15588 * ext/gio/gstgiobasesink.h:
15589 * ext/gio/gstgiobasesrc.c: (gst_gio_base_src_get_size):
15590 * ext/gio/gstgiobasesrc.h:
15591 * ext/gio/gstgiosink.c: (gst_gio_sink_start):
15592 * ext/gio/gstgiosink.h:
15593 * ext/gio/gstgiosrc.h:
15594 * ext/gio/gstgiostreamsink.h:
15595 * ext/gio/gstgiostreamsrc.h:
15596 * tests/check/pipelines/gio.c:
15597 Update to latest API changes in GLib/GIO and require at least
15598 gio-2.0 2.15.0 for this.
15599 * ext/gio/Makefile.am:
15600 Add GST_PLUGIN_LDFLAGS to LDFLAGS.
15602 2007-12-29 16:23:23 +0000 Sebastian Dröge <slomo@circular-chaos.org>
15604 ext/libvisual/visual.c: Fix 'xyz may be used uninitialized' compiler warnings caused by broken g_assert_not_reached()...
15605 Original commit message from CVS:
15606 * ext/libvisual/visual.c: (gst_visual_chain):
15607 Fix 'xyz may be used uninitialized' compiler warnings caused
15608 by broken g_assert_not_reached() macro in GLib-2.15.x and don't
15609 abort() in any case but properly report the error.
15611 2007-12-28 09:00:27 +0000 Wim Taymans <wim.taymans@gmail.com>
15613 gst/playback/gstplaybin2.c: Code cleanups.
15614 Original commit message from CVS:
15615 * gst/playback/gstplaybin2.c: (gst_play_bin_class_init),
15616 (gst_play_bin_finalize), (gst_play_bin_set_uri),
15617 (gst_play_bin_set_suburi), (gst_play_bin_set_property),
15618 (gst_play_bin_get_property), (pad_removed_cb), (drained_cb),
15619 (autoplug_select_cb), (activate_group), (deactivate_group),
15620 (setup_next_source), (save_current_group),
15621 (gst_play_bin_change_state):
15623 Remove next-uri, we can use the uri property just fine.
15625 Unref uridecodebin when switching.
15626 Fix going to READY.
15627 * gst/playback/gstplaysink.c: (gst_play_sink_class_init),
15628 (gst_play_sink_init), (gst_play_sink_dispose),
15629 (gst_play_sink_finalize), (gst_play_sink_vis_unblocked),
15630 (gst_play_sink_vis_blocked), (gst_play_sink_set_video_sink),
15631 (gst_play_sink_set_audio_sink), (gst_play_sink_set_vis_plugin),
15632 (gst_play_sink_set_property), (gst_play_sink_get_property),
15633 (gen_video_chain), (gen_text_element), (gen_audio_chain),
15634 (gen_vis_element), (gst_play_sink_get_mode),
15635 (gst_play_sink_set_mode), (gst_play_sink_set_flags),
15636 (gst_play_sink_get_flags), (gst_play_sink_request_pad),
15637 (gst_play_sink_release_pad), (gst_play_sink_send_event_to_sink),
15638 (gst_play_sink_change_state):
15639 * gst/playback/gstplaysink.h:
15640 Add some locking to make things threadsafe.
15641 * gst/playback/test7.c: (about_to_finish_cb):
15644 2007-12-22 12:06:47 +0000 Tim-Philipp Müller <tim@centricular.net>
15646 gst/videoscale/gstvideoscale.c: Don't claim to be able to handle/transform caps that can't really be handled by the c...
15647 Original commit message from CVS:
15648 * gst/videoscale/gstvideoscale.c: (gst_video_scale_set_property),
15649 (gst_video_scale_get_property), (gst_video_scale_transform_caps),
15650 (gst_video_scale_transform):
15651 Don't claim to be able to handle/transform caps that can't really
15652 be handled by the currently selected scaling method (here: RGB or
15653 packed YUV with 4-tap method). Also add locking to method property.
15654 * tests/check/pipelines/simple-launch-lines.c: (setup_pipeline),
15655 (test_basetransform_based):
15656 Some test pipelines for the above (not entirely valgrind clean yet
15659 2007-12-22 05:19:00 +0000 David Schleef <ds@schleef.org>
15661 gst-libs/gst/video/video.*: Add additional RGBA and RGB-24 video formats.
15662 Original commit message from CVS:
15663 * gst-libs/gst/video/video.c:
15664 * gst-libs/gst/video/video.h:
15665 Add additional RGBA and RGB-24 video formats.
15667 2007-12-21 22:46:56 +0000 Tim-Philipp Müller <tim@centricular.net>
15669 tests/check/: Don't use GST_PLUGIN_DEFINE_STATIC, it's not portable and will be deprecated in the future (see #498924).
15670 Original commit message from CVS:
15671 * tests/check/elements/playbin.c: (test_sink_usage_video_only_stream),
15672 (test_suburi_error_unknowntype), (test_suburi_error_invalidfile),
15673 (test_suburi_error_wrongproto), (test_missing_primary_decoder):
15674 * tests/check/libs/cddabasesrc.c: (GST_START_TEST), (GST_START_TEST),
15675 (cddabasesrc_suite):
15676 Don't use GST_PLUGIN_DEFINE_STATIC, it's not portable and will be
15677 deprecated in the future (see #498924).
15679 2007-12-21 22:26:47 +0000 Tim-Philipp Müller <tim@centricular.net>
15681 gst/playback/gststreamselector.c: Don't leak event.
15682 Original commit message from CVS:
15683 * gst/playback/gststreamselector.c: (gst_selector_pad_event):
15686 2007-12-20 19:43:25 +0000 Thijs Vermeir <thijsvermeir@gmail.com>
15688 gst-libs/gst/riff/riff-read.c: Use GST_ROUND_UP_2 macro
15689 Original commit message from CVS:
15690 * gst-libs/gst/riff/riff-read.c:
15691 Use GST_ROUND_UP_2 macro
15693 2007-12-20 17:13:37 +0000 Tim-Philipp Müller <tim@centricular.net>
15695 gst/playback/.cvsignore: Ignore more.
15696 Original commit message from CVS:
15697 * gst/playback/.cvsignore:
15700 2007-12-20 10:41:29 +0000 Tim-Philipp Müller <tim@centricular.net>
15702 Make switching off of subtitles work. To avoid all kind of problems with unlinking of the subtitle input, we just kee...
15703 Original commit message from CVS:
15704 * ext/pango/gsttextoverlay.c: (gst_text_overlay_class_init):
15705 * gst/playback/gstplaybasebin.c: (set_subtitles_visible),
15706 (set_active_source):
15707 * gst/playback/gstplaybasebin.h:
15708 * gst/playback/gstplaybin.c: (gst_play_bin_class_init),
15709 (setup_sinks), (playbin_set_subtitles_visible):
15710 Make switching off of subtitles work. To avoid all kind of
15711 problems with unlinking of the subtitle input, we just keep
15712 the subtitle inputs linked as they are and tell textoverlay
15713 not to render them. Fixes #373011.
15714 Other subtitle switching issues (esp. when there are both
15715 external and in-stream subtitles) remain. They'll be solved
15718 2007-12-18 16:21:35 +0000 Wim Taymans <wim.taymans@gmail.com>
15720 gst/playback/gststreamselector.c: Init the pad segment too.
15721 Original commit message from CVS:
15722 * gst/playback/gststreamselector.c: (gst_selector_pad_init):
15723 Init the pad segment too.
15725 2007-12-18 15:56:51 +0000 Wim Taymans <wim.taymans@gmail.com>
15727 gst-libs/gst/audio/gstaudiosink.c: Improve debug output.
15728 Original commit message from CVS:
15729 * gst-libs/gst/audio/gstaudiosink.c: (audioringbuffer_thread_func),
15730 (gst_audioringbuffer_open_device),
15731 (gst_audioringbuffer_close_device), (gst_audioringbuffer_acquire),
15732 (gst_audioringbuffer_release), (gst_audioringbuffer_start),
15733 (gst_audioringbuffer_pause), (gst_audioringbuffer_stop),
15734 (gst_audio_sink_create_ringbuffer):
15735 Improve debug output.
15736 * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_start),
15737 (gst_ring_buffer_pause), (gst_ring_buffer_delay):
15738 Prevent some functions from doing things and failing when the
15739 ringbuffer is not yet acquired.
15741 2007-12-18 15:32:49 +0000 Sebastian Dröge <slomo@circular-chaos.org>
15743 gst-libs/gst/interfaces/interfaces.h: Also remove interfaces.h from CVS as it is not needed anymore.
15744 Original commit message from CVS:
15745 * gst-libs/gst/interfaces/interfaces.h:
15746 Also remove interfaces.h from CVS as it is not needed anymore.
15748 2007-12-18 15:20:12 +0000 Sebastian Dröge <slomo@circular-chaos.org>
15750 gst-libs/gst/interfaces/Makefile.am: interfaces.h is not used anymore so remove it from the build process.
15751 Original commit message from CVS:
15752 * gst-libs/gst/interfaces/Makefile.am:
15753 interfaces.h is not used anymore so remove it from the build
15756 2007-12-18 01:01:23 +0000 David Schleef <ds@schleef.org>
15758 gst/videotestsrc/gstvideotestsrc.*: Add a "blink" pattern. Turn on the pain. Apologies. It's useful for testing ve...
15759 Original commit message from CVS:
15760 * gst/videotestsrc/gstvideotestsrc.c:
15761 * gst/videotestsrc/gstvideotestsrc.h:
15762 Add a "blink" pattern. Turn on the pain. Apologies. It's useful
15763 for testing vertical refresh synchronization.
15765 2007-12-18 00:13:26 +0000 David Schleef <ds@schleef.org>
15767 Add new GstVideFormat enum and write a bunch of helper functions based around it.
15768 Original commit message from CVS:
15769 * docs/libs/gst-plugins-base-libs-sections.txt:
15770 * gst-libs/gst/video/video.c:
15771 * gst-libs/gst/video/video.h:
15772 Add new GstVideFormat enum and write a bunch of helper functions
15775 2007-12-17 23:41:14 +0000 Tim-Philipp Müller <tim@centricular.net>
15777 Makefile.am: Use new common/win32.mak.
15778 Original commit message from CVS:
15780 Use new common/win32.mak.
15782 2007-12-17 16:44:51 +0000 Wim Taymans <wim.taymans@gmail.com>
15784 gst-libs/gst/audio/gstbaseaudiosrc.c: Add debug info.
15785 Original commit message from CVS:
15786 * gst-libs/gst/audio/gstbaseaudiosrc.c:
15787 (gst_base_audio_src_create), (gst_base_audio_src_change_state):
15789 When going from PLAYING to PAUSED, pause the ringbuffer before calling
15790 the parent state change function, just like the audiosink, because the
15791 parent waits for the element to finish its processing before completing
15792 the state change. This makes going to PAUSED a lot snappier.
15793 When going from READY to PAUSED, don't allow the ringbuffer to start
15796 2007-12-17 00:01:00 +0000 Edward Hervey <bilboed@bilboed.com>
15798 gst-libs/gst/riff/riff-media.c: Yet another fix for broken software that produce files with an empty blockalign field...
15799 Original commit message from CVS:
15800 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
15801 Yet another fix for broken software that produce files with an empty
15802 blockalign field. Instead of completely failing, make a second attempt
15803 at guessing the width/depth by looking at strf->size.
15805 2007-12-16 23:52:58 +0000 Tim-Philipp Müller <tim@centricular.net>
15807 gst-libs/gst/: Turn a few g_assert_not_reached() into g_return_val_if_reached() to avoid compiler warnings (#503930).
15808 Original commit message from CVS:
15809 * gst-libs/gst/cdda/gstcddabasesrc.c: (gst_cdda_base_src_do_seek),
15810 (gst_cdda_base_src_handle_track_seek), (gst_cdda_base_src_create):
15811 * gst-libs/gst/pbutils/install-plugins.c:
15812 (gst_install_plugins_spawn_child), (gst_install_plugins_supported):
15813 * gst-libs/gst/pbutils/missing-plugins.c:
15814 (gst_missing_plugin_message_get_installer_detail),
15815 (gst_missing_encoder_installer_detail_new):
15816 * gst-libs/gst/rtsp/gstrtspconnection.c: (gst_rtsp_connection_send):
15817 * gst-libs/gst/rtsp/gstrtspmessage.c: (gst_rtsp_message_unset):
15818 Turn a few g_assert_not_reached() into g_return_val_if_reached() to
15819 avoid compiler warnings (#503930).
15821 2007-12-16 23:46:16 +0000 Edward Hervey <bilboed@bilboed.com>
15823 gst-libs/gst/riff/riff-media.c: Be apologetic of software that use the 'jpeg' instead of 'JPEG' FOURCC for jpeg video...
15824 Original commit message from CVS:
15825 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
15826 Be apologetic of software that use the 'jpeg' instead of 'JPEG' FOURCC
15827 for jpeg video streams.
15828 Add the 'avc1'/'AVC1' fourcc mapping for h264, same software-comment as
15829 for the above modification.
15831 2007-12-15 17:27:48 +0000 Tim-Philipp Müller <tim@centricular.net>
15833 gst-libs/gst/interfaces/xoverlay.c: More guards (we don't want klass to end up being NULL).
15834 Original commit message from CVS:
15835 * gst-libs/gst/interfaces/xoverlay.c: (gst_x_overlay_expose),
15836 (gst_x_overlay_handle_events):
15837 More guards (we don't want klass to end up being NULL).
15839 2007-12-15 03:40:34 +0000 Sebastian Dröge <slomo@circular-chaos.org>
15841 Use new gst_base_transform_set_gap_aware() function as volume correctly handles GST_BUFFER_FLAG_GAP. Require core 0.1...
15842 Original commit message from CVS:
15844 * gst/volume/gstvolume.c: (gst_volume_init):
15845 Use new gst_base_transform_set_gap_aware() function as volume
15846 correctly handles GST_BUFFER_FLAG_GAP. Require core 0.10.15.1
15849 2007-12-14 19:06:24 +0000 Wim Taymans <wim.taymans@gmail.com>
15851 tests/examples/seek/seek.c: Don't go to READY on EOS as this avoids testing of seeking and restarting after EOS, use ...
15852 Original commit message from CVS:
15853 * tests/examples/seek/seek.c: (msg_segment_done), (main):
15854 Don't go to READY on EOS as this avoids testing of seeking and
15855 restarting after EOS, use the stop button when you want to READY.
15856 Don't try to do a flushing seek in segment-done, it does not make
15857 sense to use this for gapless playback and is not needed.
15859 2007-12-14 18:46:12 +0000 Wim Taymans <wim.taymans@gmail.com>
15861 gst/playback/gstqueue2.c: Use separate timers for input and output rates.
15862 Original commit message from CVS:
15863 * gst/playback/gstqueue2.c: (gst_queue_init), (gst_queue_finalize),
15864 (reset_rate_timer), (update_in_rates), (update_out_rates),
15865 (gst_queue_locked_enqueue), (gst_queue_locked_dequeue),
15866 (gst_queue_chain), (gst_queue_loop):
15867 Use separate timers for input and output rates.
15868 Pause measuring the output rate when we block for more data.
15871 2007-12-14 16:23:06 +0000 Christian Schaller <uraeus@gnome.org>
15873 * gst/speexresample/Makefile.am:
15874 update spec file and add two missing files for disting
15875 Original commit message from CVS:
15876 update spec file and add two missing files for disting
15878 2007-12-14 09:24:55 +0000 Wim Taymans <wim.taymans@gmail.com>
15880 gst/playback/gstqueue2.c: Pause the timer to measure the input rate when we block because the queue is filled. See #5...
15881 Original commit message from CVS:
15882 * gst/playback/gstqueue2.c: (gst_queue_chain):
15883 Pause the timer to measure the input rate when we block because the
15884 queue is filled. See #503262.
15886 2007-12-13 15:54:00 +0000 Peter Kjellerstedt <pkj@axis.com>
15888 gst-libs/gst/rtsp/gstrtspconnection.c: Close control sockets. Fixes #503440.
15889 Original commit message from CVS:
15890 Patch by: Peter Kjellerstedt <pkj at axis com>
15891 * gst-libs/gst/rtsp/gstrtspconnection.c:
15892 (gst_rtsp_connection_free):
15893 Close control sockets. Fixes #503440.
15895 2007-12-13 12:31:38 +0000 Wim Taymans <wim.taymans@gmail.com>
15897 gst/playback/gstdecodebin2.c: Expose the right pad in the right place with the right element.
15898 Original commit message from CVS:
15899 * gst/playback/gstdecodebin2.c: (analyze_new_pad), (connect_pad):
15900 Expose the right pad in the right place with the right element.
15902 2007-12-13 11:40:10 +0000 Tim-Philipp Müller <tim@centricular.net>
15904 gst-libs/gst/pbutils/descriptions.c: Add description for 'private' dts caps (who come up with that name?).
15905 Original commit message from CVS:
15906 * gst-libs/gst/pbutils/descriptions.c: (formats):
15907 Add description for 'private' dts caps (who come up with that name?).
15909 2007-12-13 10:10:35 +0000 Tim-Philipp Müller <tim@centricular.net>
15911 Makefile.am: Add check-exports target and run it with 'make check'.
15912 Original commit message from CVS:
15914 Add check-exports target and run it with 'make check'.
15916 Be stricter about what we export in our libraries: change regexp so that
15917 we only export _gst_foo(), but not __gst_foo().
15918 * gst-libs/gst/cdda/base64.h: (rfc822_binary):
15919 * gst-libs/gst/cdda/sha1.h: (sha_init), (sha_update), (sha_final):
15920 Change internal functions to __gst_foo so they dont' get exported.
15921 * win32/common/libgstaudio.def:
15922 Add missing symbols.
15924 2007-12-11 21:18:57 +0000 David Schleef <ds@schleef.org>
15927 ChangeLog: remove conflict markers
15928 Original commit message from CVS:
15929 ChangeLog: remove conflict markers
15931 2007-12-11 17:14:13 +0000 Tim-Philipp Müller <tim@centricular.net>
15933 ext/gnomevfs/: Use gst_tag_freeform_string_to_utf8() here, which also takes into account any character sets specified...
15934 Original commit message from CVS:
15935 * ext/gnomevfs/Makefile.am:
15936 * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_unicodify):
15937 Use gst_tag_freeform_string_to_utf8() here, which also takes
15938 into account any character sets specified by the user via
15939 environment variables.
15941 2007-12-10 15:21:41 +0000 Wim Taymans <wim.taymans@gmail.com>
15943 gst/audioconvert/Makefile.am: Also link to libm.
15944 Original commit message from CVS:
15945 * gst/audioconvert/Makefile.am:
15948 2007-12-10 15:13:55 +0000 Wim Taymans <wim.taymans@gmail.com>
15950 gst-libs/gst/riff/riff-media.c: No need for floating point operations here. avoids having to link against the math li...
15951 Original commit message from CVS:
15952 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
15953 No need for floating point operations here. avoids having to link
15954 against the math library too.
15956 2007-12-10 11:16:25 +0000 Tim-Philipp Müller <tim@centricular.net>
15958 Add one or two missing formats. Generate ADPCM description dynamically depending on layout/format.
15959 Original commit message from CVS:
15960 * gst-libs/gst/pbutils/descriptions.c: (formats),
15961 (format_info_get_desc):
15962 * tests/check/libs/pbutils.c: (GST_START_TEST), (caps_strings),
15964 Add one or two missing formats. Generate ADPCM description
15965 dynamically depending on layout/format.
15967 2007-12-09 04:28:38 +0000 Sebastian Dröge <slomo@circular-chaos.org>
15969 configure.ac: Don't define GST_DISABLE_DEPRECATED for releases. Fixes #498181.
15970 Original commit message from CVS:
15972 Don't define GST_DISABLE_DEPRECATED for releases. Fixes #498181.
15974 2007-12-08 18:38:39 +0000 Robin Stocker <robin.stocker@gmx.ch>
15976 gst/subparse/gstsubparse.c: Some .srt files start with chunk number 0 and not chunk number 1, recognise and accept th...
15977 Original commit message from CVS:
15978 Patch by: Robin Stocker <robin dot stocker at gmx dot ch>
15979 * gst/subparse/gstsubparse.c: (gst_sub_parse_data_format_autodetect):
15980 Some .srt files start with chunk number 0 and not chunk number 1,
15981 recognise and accept those as well (fixes #502497).
15982 * tests/check/elements/subparse.c: (srt_input), (srt_input0),
15984 Add unit test for the above.
15986 2007-12-06 12:08:21 +0000 Wim Taymans <wim.taymans@gmail.com>
15988 gst/playback/gstplay-enum.*: Add missing files.
15989 Original commit message from CVS:
15990 * gst/playback/gstplay-enum.c:
15991 (register_gst_autoplug_select_result),
15992 (gst_autoplug_select_result_get_type):
15993 * gst/playback/gstplay-enum.h:
15996 2007-12-05 17:11:48 +0000 Wim Taymans <wim.taymans@gmail.com>
15998 gst/playback/Makefile.am: Group decodebin2 and uridecodebin into the same plugin so that they can share the GEnumType.
15999 Original commit message from CVS:
16000 * gst/playback/Makefile.am:
16001 Group decodebin2 and uridecodebin into the same plugin so that they
16002 can share the GEnumType.
16003 * gst/playback/gstdecodebin2.c: (_gst_array_accumulator),
16004 (_gst_select_accumulator), (gst_decode_bin_class_init),
16005 (gst_decode_bin_init), (gst_decode_bin_autoplug_sort),
16006 (gst_decode_bin_autoplug_select), (gst_decode_bin_autoplug_add),
16007 (analyze_new_pad), (connect_pad), (gst_decode_bin_plugin_init):
16008 Add signal to sort factories instead of the more awkward autoplug-select
16010 Modify autoplug_select so that we can try, skip or expose the
16011 autopluggin of an element on a pad.
16012 * gst/playback/gstfactorylists.c: (compare_ranks),
16013 (decoders_filter), (sinks_filter), (gst_factory_list_is_type),
16014 (element_filter), (gst_factory_list_get_elements),
16015 (gst_factory_list_debug), (gst_factory_list_filter):
16016 * gst/playback/gstfactorylists.h:
16017 Simplify the API, allow getting elements based on mask.
16018 * gst/playback/gstplay-marshal.list:
16019 Add some more marshallers.
16020 * gst/playback/gstplaybin2.c: (init_group), (gst_play_bin_init),
16021 (gst_play_bin_finalize), (pad_removed_cb), (autoplug_factories_cb),
16022 (autoplug_select_cb), (activate_group):
16023 Add support for managing non-raw sinks by providing a custom element and
16024 sink list to decodebin2.
16025 Try to plug non-raw sinks when decodebin2 using autoplug-select of
16027 * gst/playback/gstplaysink.c: (gen_video_chain), (gen_audio_chain),
16028 (gst_play_sink_set_mode), (gst_play_sink_request_pad):
16029 * gst/playback/gstplaysink.h:
16030 Add support for raw and non-raw sinks.
16031 Add support to force sinks selected by playbin2.
16032 Don't plug raw converters for non-raw sinks.
16033 * gst/playback/gsturidecodebin.c: (_gst_array_accumulator),
16034 (_gst_select_accumulator), (gst_uri_decode_bin_class_init),
16035 (proxy_autoplug_select_signal), (gst_uri_decode_bin_plugin_init),
16037 Use right accumulators.
16040 2007-12-03 13:47:00 +0000 Wim Taymans <wim.taymans@gmail.com>
16042 gst-libs/gst/rtp/gstbasertppayload.c: Use runnning time as the base time instead of the timestamp.
16043 Original commit message from CVS:
16044 * gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_push):
16045 Use runnning time as the base time instead of the timestamp.
16046 Spotted by Saur on IRC.
16048 2007-12-03 11:32:30 +0000 Edward Hervey <bilboed@bilboed.com>
16050 gst-libs/gst/riff/riff-media.c: Add 'WVC1' codec mapping for Windows Media VC-1 video codec.
16051 Original commit message from CVS:
16052 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
16053 Add 'WVC1' codec mapping for Windows Media VC-1 video codec.
16055 2007-12-03 10:58:14 +0000 Wim Taymans <wim.taymans@gmail.com>
16057 ext/ogg/gstoggdemux.c: If we find a new serial number but it does not contain a BOS page, make sure we initialize the...
16058 Original commit message from CVS:
16059 * ext/ogg/gstoggdemux.c: (gst_ogg_demux_bisect_forward_serialno),
16060 (gst_ogg_demux_read_chain):
16061 If we find a new serial number but it does not contain a BOS page, make
16062 sure we initialize the chain to NULL because else we will try to scan it
16063 and crash. Fixes #500763
16065 2007-11-30 17:47:15 +0000 Wim Taymans <wim.taymans@gmail.com>
16067 gst/playback/: Refactor some common code to filter factories and check caps compat.
16068 Original commit message from CVS:
16069 * gst/playback/Makefile.am:
16070 * gst/playback/gstfactorylists.c: (compare_ranks), (print_feature),
16071 (get_feature_array), (decoders_filter), (sinks_filter),
16072 (gst_factory_list_get_decoders), (gst_factory_list_get_sinks),
16073 (gst_factory_list_filter):
16074 * gst/playback/gstfactorylists.h:
16075 Refactor some common code to filter factories and check caps compat.
16076 * gst/playback/gstdecodebin.c:
16077 * gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init),
16078 (gst_decode_bin_init), (gst_decode_bin_dispose),
16079 (gst_decode_bin_autoplug_continue),
16080 (gst_decode_bin_autoplug_factories),
16081 (gst_decode_bin_autoplug_select), (analyze_new_pad),
16082 (find_compatibles):
16083 * gst/playback/gstplaybin.c:
16084 * gst/playback/gstplaybin2.c: (gst_play_bin_class_init),
16085 (gst_play_bin_init), (gst_play_bin_finalize),
16086 (autoplug_factories_cb), (activate_group):
16087 * gst/playback/gstqueue2.c:
16088 * gst/playback/gsturidecodebin.c: (proxy_unknown_type_signal),
16089 (proxy_autoplug_continue_signal),
16090 (proxy_autoplug_factories_signal), (proxy_autoplug_select_signal),
16091 (proxy_drained_signal):
16092 Add some more debug info and use factor filtering code.
16094 2007-11-26 13:19:46 +0000 Julien Moutte <julien@moutte.net>
16096 configure.ac: Add QuickTime Wrapper plug-in.
16097 Original commit message from CVS:
16098 2007-11-26 Julien Moutte <julien@fluendo.com>
16099 * configure.ac: Add QuickTime Wrapper plug-in.
16100 * gst/speexresample/gstspeexresample.c:
16101 (gst_speex_resample_push_drain), (gst_speex_resample_process): Fix
16102 build on Mac OS X Leopard. Incorrect printf format arguments.
16104 * sys/qtwrapper/Makefile.am:
16105 * sys/qtwrapper/audiodecoders.c:
16106 (qtwrapper_audio_decoder_base_init),
16107 (qtwrapper_audio_decoder_class_init),
16108 (qtwrapper_audio_decoder_init),
16109 (clear_AudioStreamBasicDescription), (fill_indesc_mp3),
16110 (fill_indesc_aac), (fill_indesc_samr), (fill_indesc_generic),
16111 (make_samr_magic_cookie), (open_decoder),
16112 (qtwrapper_audio_decoder_sink_setcaps), (process_buffer_cb),
16113 (qtwrapper_audio_decoder_chain),
16114 (qtwrapper_audio_decoder_sink_event),
16115 (qtwrapper_audio_decoders_register):
16116 * sys/qtwrapper/codecmapping.c: (audio_caps_from_string),
16118 * sys/qtwrapper/codecmapping.h:
16119 * sys/qtwrapper/imagedescription.c: (image_description_for_avc1),
16120 (image_description_for_mp4v), (image_description_from_stsd_buffer),
16121 (image_description_from_codec_data):
16122 * sys/qtwrapper/imagedescription.h:
16123 * sys/qtwrapper/qtutils.c: (get_name_info_from_component),
16124 (get_output_info_from_component), (dump_avcc_atom),
16125 (dump_image_description), (dump_codec_decompress_params),
16126 (addSInt32ToDictionary), (dump_cvpixel_buffer),
16127 (DestroyAudioBufferList), (AllocateAudioBufferList):
16128 * sys/qtwrapper/qtutils.h:
16129 * sys/qtwrapper/qtwrapper.c: (plugin_init):
16130 * sys/qtwrapper/qtwrapper.h:
16131 * sys/qtwrapper/videodecoders.c:
16132 (qtwrapper_video_decoder_base_init),
16133 (qtwrapper_video_decoder_class_init),
16134 (qtwrapper_video_decoder_init), (qtwrapper_video_decoder_finalize),
16135 (fill_image_description), (new_image_description), (close_decoder),
16136 (open_decoder), (qtwrapper_video_decoder_sink_setcaps),
16137 (decompressCb), (qtwrapper_video_decoder_chain),
16138 (qtwrapper_video_decoder_sink_event),
16139 (qtwrapper_video_decoders_register): Initial import of QuickTime
16140 wrapper jointly developped by Songbird authors (Pioneers of the
16141 Inevitable) and Fluendo.
16143 2007-11-26 12:25:55 +0000 Stefan Kost <ensonic@users.sourceforge.net>
16145 gst/: Add GAP-flag support.
16146 Original commit message from CVS:
16147 * gst/audiotestsrc/gstaudiotestsrc.c:
16148 * gst/volume/gstvolume.c:
16149 * gst/volume/gstvolume.h:
16150 Add GAP-flag support.
16152 2007-11-26 08:43:25 +0000 Sebastian Dröge <slomo@circular-chaos.org>
16154 gst/speexresample/: Update speex resampler to latest SVN. We're now down to only the changes noted in README again.
16155 Original commit message from CVS:
16156 * gst/speexresample/README:
16157 * gst/speexresample/arch.h:
16158 * gst/speexresample/resample.c: (resampler_basic_direct_single),
16159 (resampler_basic_direct_double),
16160 (resampler_basic_interpolate_single),
16161 (resampler_basic_interpolate_double),
16162 (speex_resampler_process_native), (speex_resampler_process_float),
16163 (speex_resampler_process_int),
16164 (speex_resampler_process_interleaved_float),
16165 (speex_resampler_process_interleaved_int),
16166 (speex_resampler_get_input_latency),
16167 (speex_resampler_get_output_latency):
16168 * gst/speexresample/speex_resampler.h:
16169 Update speex resampler to latest SVN. We're now down to only the
16170 changes noted in README again.
16171 * gst/speexresample/speex_resampler_wrapper.h:
16172 * gst/speexresample/gstspeexresample.c:
16173 (gst_speex_resample_push_drain), (gst_speex_resample_query):
16174 Adjust to API changes.
16176 2007-11-24 15:02:01 +0000 Julien Moutte <julien@moutte.net>
16178 tests/examples/seek/seek.c: Increase the range of the rate selector as I would like to test QOS behavior at higher fo...
16179 Original commit message from CVS:
16180 2007-11-24 Julien MOUTTE <julien@moutte.net>
16181 * tests/examples/seek/seek.c: (main): Increase the range of the
16182 rate selector as I would like to test QOS behavior at higher
16183 forward and reverse playback speed like say 64x.
16185 2007-11-23 10:21:31 +0000 Sebastian Dröge <slomo@circular-chaos.org>
16187 gst/speexresample/gstspeexresample.c: Only post the latency message if we have a resampler state already.
16188 Original commit message from CVS:
16189 * gst/speexresample/gstspeexresample.c:
16190 (gst_speex_resample_update_state):
16191 Only post the latency message if we have a resampler state already.
16193 2007-11-23 10:21:11 +0000 Sebastian Dröge <slomo@circular-chaos.org>
16195 gst/audioresample/gstaudioresample.c: Implement latency query.
16196 Original commit message from CVS:
16197 * gst/audioresample/gstaudioresample.c: (gst_audioresample_init),
16198 (audioresample_query), (audioresample_query_type),
16199 (gst_audioresample_set_property):
16200 Implement latency query.
16202 2007-11-23 10:01:33 +0000 Sebastian Dröge <slomo@circular-chaos.org>
16204 gst/speexresample/gstspeexresample.c: Also post GST_MESSAGE_LATENCY if the latency changes.
16205 Original commit message from CVS:
16206 * gst/speexresample/gstspeexresample.c:
16207 (gst_speex_resample_update_state):
16208 Also post GST_MESSAGE_LATENCY if the latency changes.
16210 2007-11-23 08:48:50 +0000 Sebastian Dröge <slomo@circular-chaos.org>
16212 gst/speexresample/: Add functions to push the remaining samples and to get the latency of the resampler. These will g...
16213 Original commit message from CVS:
16214 * gst/speexresample/resample.c: (speex_resampler_get_latency),
16215 (speex_resampler_drain_float), (speex_resampler_drain_int),
16216 (speex_resampler_drain_interleaved_float),
16217 (speex_resampler_drain_interleaved_int):
16218 * gst/speexresample/speex_resampler.h:
16219 * gst/speexresample/speex_resampler_wrapper.h:
16220 Add functions to push the remaining samples and to get the latency
16221 of the resampler. These will get added to Speex SVN in this or a
16222 slightly changed form at some point too and should get merged then
16224 * gst/speexresample/gstspeexresample.c: (gst_speex_resample_init),
16225 (gst_speex_resample_init_state),
16226 (gst_speex_resample_transform_size),
16227 (gst_speex_resample_push_drain), (gst_speex_resample_event),
16228 (gst_speex_fix_output_buffer), (gst_speex_resample_process),
16229 (gst_speex_resample_query), (gst_speex_resample_query_type):
16230 Drop the prepending zeroes and output the remaining samples on EOS.
16231 Also properly implement the latency query for this. speexresample
16232 should be completely ready for production use now.
16234 2007-11-21 18:02:21 +0000 Wim Taymans <wim.taymans@gmail.com>
16236 gst-libs/gst/audio/gstbaseaudiosink.c: Our EOS time contains the base_time, _wait_eos() expects a running_time so we ...
16237 Original commit message from CVS:
16238 * gst-libs/gst/audio/gstbaseaudiosink.c:
16239 (gst_base_audio_sink_drain):
16240 Our EOS time contains the base_time, _wait_eos() expects a running_time
16241 so we have to subtract the base_time again before calling the function.
16242 This fixes an EOS regression where the base_time was added twice and EOS
16243 took longer and longer in certain situations.
16246 2007-11-21 13:04:17 +0000 Wim Taymans <wim.taymans@gmail.com>
16248 Expose methods for some object properties so that subclasses can more easily configure them.
16249 Original commit message from CVS:
16250 * docs/libs/gst-plugins-base-libs-sections.txt:
16251 * gst-libs/gst/audio/gstbaseaudiosink.c: (slave_method_get_type),
16252 (gst_base_audio_sink_set_provide_clock),
16253 (gst_base_audio_sink_get_provide_clock),
16254 (gst_base_audio_sink_set_slave_method),
16255 (gst_base_audio_sink_get_slave_method),
16256 (gst_base_audio_sink_set_property),
16257 (gst_base_audio_sink_get_property), (gst_base_audio_sink_drain),
16258 (gst_base_audio_sink_none_slaving),
16259 (gst_base_audio_sink_handle_slaving):
16260 * gst-libs/gst/audio/gstbaseaudiosink.h:
16261 Expose methods for some object properties so that subclasses can more
16262 easily configure them.
16263 Added slave method none, that completely disables slaving to the
16265 API: gst_base_audio_sink_set_provide_clock()
16266 API: gst_base_audio_sink_get_provide_clock()
16267 API: gst_base_audio_sink_set_slave_method()
16268 API: gst_base_audio_sink_get_slave_method()
16269 * gst-libs/gst/audio/gstbaseaudiosrc.c:
16270 (gst_base_audio_src_set_provide_clock),
16271 (gst_base_audio_src_get_provide_clock),
16272 (gst_base_audio_src_set_property),
16273 (gst_base_audio_src_get_property), (gst_base_audio_src_create):
16274 * gst-libs/gst/audio/gstbaseaudiosrc.h:
16275 Expose methods for some object properties so that subclasses can more
16276 easily configure them.
16277 API: gst_base_audio_src_set_provide_clock()
16278 API: gst_base_audio_src_get_provide_clock()
16280 2007-11-21 10:18:56 +0000 Sebastian Dröge <slomo@circular-chaos.org>
16282 gst/speexresample/README: Add README explaining where the resampling code was taken from and which changes were done.
16283 Original commit message from CVS:
16284 * gst/speexresample/README:
16285 Add README explaining where the resampling code was taken from
16286 and which changes were done.
16287 * gst/speexresample/resample.c: (speex_alloc), (speex_realloc),
16289 Use g_malloc() and friends instead of malloc() to achieve higher
16290 portability and define the functions inline.
16291 * gst/speexresample/speex_resampler.h:
16292 Add back some useless preprocessor stuff to keep the diff between
16293 our version and the one from the Speex SVN repository lower.
16295 2007-11-20 20:23:25 +0000 Sebastian Dröge <slomo@circular-chaos.org>
16297 gst/speexresample/gstspeexresample.c: Some small cleanup and addition of a TODO item.
16298 Original commit message from CVS:
16299 * gst/speexresample/gstspeexresample.c:
16300 (gst_speex_fix_output_buffer), (gst_speex_resample_transform):
16301 Some small cleanup and addition of a TODO item.
16303 2007-11-20 12:56:00 +0000 Sebastian Dröge <slomo@circular-chaos.org>
16305 gst/speexresample/Makefile.am: Add missing file.
16306 Original commit message from CVS:
16307 * gst/speexresample/Makefile.am:
16310 2007-11-20 07:53:56 +0000 Joe Peterson <lavajoe@gentoo.org>
16312 gst-libs/gst/sdp/gstsdpmessage.c: Fix compilation on FreeBSD (Gentoo). Fixes #498228.
16313 Original commit message from CVS:
16314 Patch by: Joe Peterson <lavajoe at gentoo dot org>
16315 * gst-libs/gst/sdp/gstsdpmessage.c:
16316 Fix compilation on FreeBSD (Gentoo). Fixes #498228.
16318 2007-11-20 07:47:27 +0000 Sebastian Dröge <slomo@circular-chaos.org>
16320 Add speexresample to the docs and while at that do a make update.
16321 Original commit message from CVS:
16322 * docs/plugins/Makefile.am:
16323 * docs/plugins/gst-plugins-bad-plugins-docs.sgml:
16324 * docs/plugins/gst-plugins-bad-plugins-sections.txt:
16325 * docs/plugins/gst-plugins-bad-plugins.args:
16326 * docs/plugins/gst-plugins-bad-plugins.signals:
16327 * docs/plugins/inspect/plugin-bz2.xml:
16328 * docs/plugins/inspect/plugin-cdxaparse.xml:
16329 * docs/plugins/inspect/plugin-dtsdec.xml:
16330 * docs/plugins/inspect/plugin-equalizer.xml:
16331 * docs/plugins/inspect/plugin-faac.xml:
16332 * docs/plugins/inspect/plugin-faad.xml:
16333 * docs/plugins/inspect/plugin-filter.xml:
16334 * docs/plugins/inspect/plugin-freeze.xml:
16335 * docs/plugins/inspect/plugin-gio.xml:
16336 * docs/plugins/inspect/plugin-gsm.xml:
16337 * docs/plugins/inspect/plugin-gstrtpmanager.xml:
16338 * docs/plugins/inspect/plugin-h264parse.xml:
16339 * docs/plugins/inspect/plugin-modplug.xml:
16340 * docs/plugins/inspect/plugin-mpeg2enc.xml:
16341 * docs/plugins/inspect/plugin-musepack.xml:
16342 * docs/plugins/inspect/plugin-musicbrainz.xml:
16343 * docs/plugins/inspect/plugin-nsfdec.xml:
16344 * docs/plugins/inspect/plugin-replaygain.xml:
16345 * docs/plugins/inspect/plugin-soundtouch.xml:
16346 * docs/plugins/inspect/plugin-spcdec.xml:
16347 * docs/plugins/inspect/plugin-spectrum.xml:
16348 * docs/plugins/inspect/plugin-speed.xml:
16349 * docs/plugins/inspect/plugin-tta.xml:
16350 * docs/plugins/inspect/plugin-videosignal.xml:
16351 * docs/plugins/inspect/plugin-xingheader.xml:
16352 * docs/plugins/inspect/plugin-xvid.xml:
16353 * gst/speexresample/gstspeexresample.h:
16354 Add speexresample to the docs and while at that do a make update.
16356 2007-11-20 07:30:30 +0000 Sebastian Dröge <slomo@circular-chaos.org>
16358 gst/speexresample/gstspeexresample.c: If the resampler gives less output samples than expected adjust the output buff...
16359 Original commit message from CVS:
16360 * gst/speexresample/gstspeexresample.c:
16361 (gst_speex_fix_output_buffer), (gst_speex_resample_process):
16362 If the resampler gives less output samples than expected
16363 adjust the output buffer and print a warning.
16365 2007-11-20 07:02:45 +0000 Sebastian Dröge <slomo@circular-chaos.org>
16367 Add resample element based on the Speex resampling algorithm.
16368 Original commit message from CVS:
16370 * gst/speexresample/arch.h:
16371 * gst/speexresample/fixed_generic.h:
16372 * gst/speexresample/gstspeexresample.c:
16373 (gst_speex_resample_base_init), (gst_speex_resample_class_init),
16374 (gst_speex_resample_init), (gst_speex_resample_start),
16375 (gst_speex_resample_stop), (gst_speex_resample_get_unit_size),
16376 (gst_speex_resample_transform_caps),
16377 (gst_speex_resample_init_state), (gst_speex_resample_update_state),
16378 (gst_speex_resample_reset_state), (gst_speex_resample_parse_caps),
16379 (gst_speex_resample_transform_size), (gst_speex_resample_set_caps),
16380 (gst_speex_resample_event), (gst_speex_resample_check_discont),
16381 (gst_speex_resample_process), (gst_speex_resample_transform),
16382 (gst_speex_resample_set_property),
16383 (gst_speex_resample_get_property), (plugin_init):
16384 * gst/speexresample/gstspeexresample.h:
16385 * gst/speexresample/resample.c: (speex_alloc), (speex_realloc),
16386 (speex_free), (compute_func), (main), (sinc), (cubic_coef),
16387 (resampler_basic_direct_single), (resampler_basic_direct_double),
16388 (resampler_basic_interpolate_single),
16389 (resampler_basic_interpolate_double), (update_filter),
16390 (speex_resampler_init), (speex_resampler_init_frac),
16391 (speex_resampler_destroy), (speex_resampler_process_native),
16392 (speex_resampler_process_float), (speex_resampler_process_int),
16393 (speex_resampler_process_interleaved_float),
16394 (speex_resampler_process_interleaved_int),
16395 (speex_resampler_set_rate), (speex_resampler_get_rate),
16396 (speex_resampler_set_rate_frac), (speex_resampler_get_ratio),
16397 (speex_resampler_set_quality), (speex_resampler_get_quality),
16398 (speex_resampler_set_input_stride),
16399 (speex_resampler_get_input_stride),
16400 (speex_resampler_set_output_stride),
16401 (speex_resampler_get_output_stride), (speex_resampler_skip_zeros),
16402 (speex_resampler_reset_mem), (speex_resampler_strerror):
16403 * gst/speexresample/speex_resampler.h:
16404 * gst/speexresample/speex_resampler_float.c:
16405 * gst/speexresample/speex_resampler_int.c:
16406 * gst/speexresample/speex_resampler_wrapper.h:
16407 Add resample element based on the Speex resampling algorithm.
16409 2007-11-19 12:30:22 +0000 Sebastian Dröge <slomo@circular-chaos.org>
16411 tests/check/libs/fft.c: Fix scaling to really have dB instead of something else.
16412 Original commit message from CVS:
16413 * tests/check/libs/fft.c: (GST_START_TEST):
16414 Fix scaling to really have dB instead of something else.
16416 2007-11-19 12:08:16 +0000 Julien Moutte <julien@moutte.net>
16418 tests/examples/seek/seek.c: There's a nice macro to check
16419 Original commit message from CVS:
16420 2007-11-19 Julien MOUTTE <julien@moutte.net>
16421 * tests/examples/seek/seek.c: (main): There's a nice macro to
16423 GTK version, use it.
16425 2007-11-19 11:59:20 +0000 Julien Moutte <julien@moutte.net>
16427 tests/examples/seek/seek.c: Try to support stable version of GTK.
16428 Original commit message from CVS:
16429 2007-11-19 Julien MOUTTE <julien@moutte.net>
16430 * tests/examples/seek/seek.c: (main): Try to support stable version
16433 2007-11-17 15:25:15 +0000 Stefan Kost <ensonic@users.sourceforge.net>
16435 gst/playback/: Fix the build + little README update.
16436 Original commit message from CVS:
16437 * gst/playback/README:
16438 * gst/playback/test7.c:
16439 Fix the build + little README update.
16441 2007-11-16 16:02:45 +0000 Wim Taymans <wim.taymans@gmail.com>
16443 tests/examples/seek/seek.c: Add playbin2 seek pipeline.
16444 Original commit message from CVS:
16445 * tests/examples/seek/seek.c: (make_playerbin2_pipeline), (main):
16446 Add playbin2 seek pipeline.
16448 2007-11-16 15:44:48 +0000 Wim Taymans <wim.taymans@gmail.com>
16450 gst/playback/: Add playbin2.
16451 Original commit message from CVS:
16452 * gst/playback/Makefile.am:
16453 * gst/playback/gstplayback.c: (plugin_init):
16454 * gst/playback/test7.c: (update_scale), (warning_cb), (error_cb),
16455 (eos_cb), (about_to_finish_cb), (main):
16457 Added gapless playback example.
16458 * gst/playback/gstplaybasebin.c:
16459 * gst/playback/gstplaybasebin.h:
16460 * gst/playback/gstplaybin.c: (gst_play_bin_plugin_init):
16461 * gst/playback/gstqueue2.c:
16462 * gst/playback/test.c:
16463 * gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_class_init),
16465 * gst/playback/gststreaminfo.h:
16467 * gst/playback/gstplaybin2.c: (gst_play_bin_get_type),
16468 (gst_play_bin_class_init), (init_group), (gst_play_bin_init),
16469 (gst_play_bin_dispose), (gst_play_bin_set_uri),
16470 (gst_play_bin_set_suburi), (gst_play_bin_set_property),
16471 (gst_play_bin_get_property), (gst_play_bin_handle_message),
16472 (pad_added_cb), (pad_removed_cb), (no_more_pads_cb), (perform_eos),
16473 (drained_cb), (unlink_group), (activate_group),
16474 (setup_next_source), (gst_play_bin_change_state),
16475 (gst_play_bin2_plugin_init):
16476 Added raw first version of playbin2. Does chained oggs and gapless
16477 playback fine. No support for raw sinks yet. No visualisations or
16479 * gst/playback/gstplaysink.c: (gst_play_sink_get_type),
16480 (gst_play_sink_class_init), (gst_play_sink_init),
16481 (gst_play_sink_dispose), (gst_play_sink_vis_unblocked),
16482 (gst_play_sink_vis_blocked), (gst_play_sink_set_video_sink),
16483 (gst_play_sink_set_audio_sink), (gst_play_sink_set_vis_plugin),
16484 (gst_play_sink_set_property), (gst_play_sink_get_property),
16485 (post_missing_element_message), (free_chain), (add_chain),
16486 (activate_chain), (gen_video_chain), (gen_text_element),
16487 (gen_audio_chain), (gen_vis_element), (gst_play_sink_get_mode),
16488 (gst_play_sink_set_mode), (gst_play_sink_request_pad),
16489 (gst_play_sink_release_pad), (gst_play_sink_send_event_to_sink),
16490 (gst_play_sink_send_event), (gst_play_sink_change_state):
16491 * gst/playback/gstplaysink.h:
16492 Added Element that abstracts the sinks and their pipelines for playbin2.
16494 2007-11-16 15:05:07 +0000 Wim Taymans <wim.taymans@gmail.com>
16496 gst/playback/gststreamselector.*: Improve streamselector, make it select and unselect the current pad more intelligen...
16497 Original commit message from CVS:
16498 * gst/playback/gststreamselector.c: (gst_selector_pad_get_type),
16499 (gst_selector_pad_class_init), (gst_selector_pad_init),
16500 (gst_selector_pad_finalize), (gst_selector_pad_reset),
16501 (gst_selector_pad_get_linked_pads), (gst_selector_pad_event),
16502 (gst_selector_pad_getcaps), (gst_selector_pad_bufferalloc),
16503 (gst_selector_pad_chain), (gst_stream_selector_get_type),
16504 (gst_stream_selector_base_init), (gst_stream_selector_class_init),
16505 (gst_stream_selector_init), (gst_stream_selector_set_property),
16506 (gst_stream_selector_get_linked_pad),
16507 (gst_stream_selector_getcaps),
16508 (gst_stream_selector_is_active_sinkpad),
16509 (gst_stream_selector_activate_sinkpad),
16510 (gst_stream_selector_get_linked_pads),
16511 (gst_stream_selector_request_new_pad),
16512 (gst_stream_selector_release_pad):
16513 * gst/playback/gststreamselector.h:
16514 Improve streamselector, make it select and unselect the current pad more
16516 Subclass GstPad for the sinkpads of the selector.
16517 Handle segments more correctly.
16518 Fix caps negotiation.
16519 Implement release_pad.
16521 2007-11-16 12:51:44 +0000 Wim Taymans <wim.taymans@gmail.com>
16523 gst/playback/gstdecodebin2.c: Add drained signal fired when decodebin finishes decoding the data.
16524 Original commit message from CVS:
16525 * gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init),
16526 (gst_decode_group_check_if_drained), (source_pad_event_probe),
16528 Add drained signal fired when decodebin finishes decoding the data.
16529 Remove deprecated STATE_DIRTY message.
16530 * gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_class_init),
16531 (unknown_type_cb), (new_decoded_pad_cb), (pad_removed_cb),
16532 (analyse_source), (proxy_drained_signal), (make_decoder),
16533 (source_new_pad), (value_list_append_structure_list),
16534 (handle_redirect_message), (handle_message):
16535 Proxy the new drained signal.
16536 Handle pad removed from decodebin.
16537 Handle redirect messages by sorting multiple redirections based on the
16540 2007-11-16 11:22:09 +0000 Tommi Myöhänen <ext-tommi.myohanen@nokia.com>
16542 gst-libs/gst/rtsp/gstrtspmessage.c: Fix leaking headers. Fixes #496761.
16543 Original commit message from CVS:
16544 Patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com>
16545 * gst-libs/gst/rtsp/gstrtspmessage.c: (gst_rtsp_message_unset):
16546 Fix leaking headers. Fixes #496761.
16548 2007-11-16 11:16:58 +0000 Tommi Myöhänen <ext-tommi.myohanen@nokia.com>
16550 sys/: Don't leak the PAR on errors. Fixes #496731.
16551 Original commit message from CVS:
16552 Patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com>
16553 * sys/ximage/ximagesink.c: (gst_ximagesink_xcontext_get),
16554 (gst_ximagesink_change_state):
16555 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xcontext_get):
16556 Don't leak the PAR on errors. Fixes #496731.
16558 2007-11-16 10:14:34 +0000 Tim-Philipp Müller <tim@centricular.net>
16560 gst-libs/gst/tag/gstid3tag.c: Add mapping for audio cd discid tags, so we can extract them from tags as well (see #34...
16561 Original commit message from CVS:
16562 * gst-libs/gst/tag/gstid3tag.c: (user_tag_matches),
16563 (gst_tag_from_id3_user_tag):
16564 Add mapping for audio cd discid tags, so we can extract
16565 them from tags as well (see #347848). Also compare identifiers
16566 in ID3v2 TXXX frames in a case-insensitive way to increase
16567 compatibility when reading tags (discid vs. DiscID vs. DiscId).
16569 2007-11-16 01:21:40 +0000 Jan Schmidt <thaytan@mad.scientist.com>
16571 gst-plugins-base.doap: Oops, fix the release name.
16572 Original commit message from CVS:
16573 * gst-plugins-base.doap:
16574 Oops, fix the release name.
16576 2007-11-16 00:44:58 +0000 Jan Schmidt <thaytan@mad.scientist.com>
16578 gst-plugins-base.doap: Add 0.10.15 release
16579 Original commit message from CVS:
16580 * gst-plugins-base.doap:
16581 Add 0.10.15 release
16583 2007-11-16 00:24:55 +0000 Jan Schmidt <thaytan@mad.scientist.com>
16585 configure.ac: Back to CVS
16586 Original commit message from CVS:
16590 === release 0.10.15 ===
16592 2007-11-16 00:14:33 +0000 Jan Schmidt <thaytan@mad.scientist.com>
16594 configure.ac: releasing 0.10.15, "No need to argue"
16595 Original commit message from CVS:
16596 === release 0.10.15 ===
16597 2007-11-15 Jan Schmidt <jan.schmidt@sun.com>
16599 releasing 0.10.15, "No need to argue"
16601 2007-11-16 00:04:24 +0000 Jan Schmidt <thaytan@mad.scientist.com>
16626 Original commit message from CVS:
16629 2007-11-15 21:40:53 +0000 Jan Schmidt <thaytan@mad.scientist.com>
16631 win32/vs6/libgstfft.dsp: Convert line endings to DOS.
16632 Original commit message from CVS:
16633 * win32/vs6/libgstfft.dsp:
16634 Convert line endings to DOS.
16636 2007-11-15 21:14:04 +0000 Sébastien Moutte <sebastien@moutte.net>
16638 win32/: Add a project file for fft plugin and remove socket based plugin which don't build from the workspace.* win32...
16639 Original commit message from CVS:
16640 * win32/vs6/gst_plugins_base.dsw:
16641 * win32/vs6/libgstfft.dsp:
16643 Add a project file for fft plugin and remove socket
16644 based plugin which don't build from the workspace.* win32/vs6/libgstaudio.dsp:
16645 * win32/vs6/libgstrtp.dsp:
16646 * win32/vs6/libgsttag.dsp:
16647 Convert line endings back to DOS.
16650 2007-11-14 12:27:13 +0000 Jan Schmidt <thaytan@mad.scientist.com>
16652 win32/vs6/: Convert line endings back to DOS
16653 Original commit message from CVS:
16654 * win32/vs6/libgstinterfaces.dsp:
16655 * win32/vs6/libgstrtsp.dsp:
16656 Convert line endings back to DOS
16658 2007-11-14 11:08:48 +0000 Jan Schmidt <thaytan@mad.scientist.com>
16660 gst-libs/gst/fft/: Don't include malloc.h which doesn't exist on Mac OSX.
16661 Original commit message from CVS:
16662 * gst-libs/gst/fft/kiss_fft_f32.h:
16663 * gst-libs/gst/fft/kiss_fft_f64.h:
16664 * gst-libs/gst/fft/kiss_fft_s16.h:
16665 * gst-libs/gst/fft/kiss_fft_s32.h:
16666 Don't include malloc.h which doesn't exist on Mac OSX.
16667 Instead, pull in glib.h and use g_malloc/g_free for
16668 consistency. Fixes: #496548
16670 2007-11-09 15:54:45 +0000 Stefan Kost <ensonic@users.sourceforge.net>
16672 gst/playback/gstdecodebin2.c: Dont leak ghostpad. Fixes #475451.
16673 Original commit message from CVS:
16674 * gst/playback/gstdecodebin2.c:
16675 Dont leak ghostpad. Fixes #475451.
16677 2007-11-09 12:21:52 +0000 Wim Taymans <wim.taymans@gmail.com>
16679 Update some more docs and comments.
16680 Original commit message from CVS:
16681 * docs/design/design-decodebin.txt:
16682 * gst/playback/gstdecodebin2.c: (analyze_new_pad):
16683 Update some more docs and comments.
16685 2007-11-07 16:47:32 +0000 Sebastian Dröge <slomo@circular-chaos.org>
16687 Require GIO >= 0.1.2 and adjust unit test for an API change.
16688 Original commit message from CVS:
16690 * tests/check/pipelines/gio.c: (GST_START_TEST):
16691 Require GIO >= 0.1.2 and adjust unit test for an API change.
16693 2007-11-07 15:18:54 +0000 Sebastian Dröge <slomo@circular-chaos.org>
16695 ext/gio/gstgio.h: Add macro to check if a stream supports seeking.
16696 Original commit message from CVS:
16697 * ext/gio/gstgio.h:
16698 Add macro to check if a stream supports seeking.
16699 * ext/gio/Makefile.am:
16700 * ext/gio/gstgiobasesink.c: (gst_gio_base_sink_base_init),
16701 (gst_gio_base_sink_class_init), (gst_gio_base_sink_init),
16702 (gst_gio_base_sink_finalize), (gst_gio_base_sink_start),
16703 (gst_gio_base_sink_stop), (gst_gio_base_sink_unlock),
16704 (gst_gio_base_sink_unlock_stop), (gst_gio_base_sink_event),
16705 (gst_gio_base_sink_render), (gst_gio_base_sink_query),
16706 (gst_gio_base_sink_set_stream):
16707 * ext/gio/gstgiobasesink.h:
16708 * ext/gio/gstgiobasesrc.c: (gst_gio_base_src_base_init),
16709 (gst_gio_base_src_class_init), (gst_gio_base_src_init),
16710 (gst_gio_base_src_finalize), (gst_gio_base_src_start),
16711 (gst_gio_base_src_stop), (gst_gio_base_src_get_size),
16712 (gst_gio_base_src_is_seekable), (gst_gio_base_src_unlock),
16713 (gst_gio_base_src_unlock_stop), (gst_gio_base_src_check_get_range),
16714 (gst_gio_base_src_create), (gst_gio_base_src_set_stream):
16715 * ext/gio/gstgiobasesrc.h:
16716 Refactor common GIO functions to GstGioBaseSink and GstGioBaseSrc
16717 base classes that only require a GInputStream or GOutputStream to
16719 * ext/gio/gstgiosink.c: (gst_gio_sink_base_init),
16720 (gst_gio_sink_class_init), (gst_gio_sink_init),
16721 (gst_gio_sink_finalize), (gst_gio_sink_start):
16722 * ext/gio/gstgiosink.h:
16723 * ext/gio/gstgiosrc.c: (gst_gio_src_base_init),
16724 (gst_gio_src_class_init), (gst_gio_src_init),
16725 (gst_gio_src_finalize), (gst_gio_src_start):
16726 * ext/gio/gstgiosrc.h:
16727 Use the newly created base classes here.
16728 * ext/gio/gstgio.c: (plugin_init):
16729 * ext/gio/gstgiostreamsink.c: (gst_gio_stream_sink_base_init),
16730 (gst_gio_stream_sink_class_init), (gst_gio_stream_sink_init),
16731 (gst_gio_stream_sink_finalize), (gst_gio_stream_sink_set_property),
16732 (gst_gio_stream_sink_get_property):
16733 * ext/gio/gstgiostreamsink.h:
16734 * ext/gio/gstgiostreamsrc.c: (gst_gio_stream_src_base_init),
16735 (gst_gio_stream_src_class_init), (gst_gio_stream_src_init),
16736 (gst_gio_stream_src_finalize), (gst_gio_stream_src_set_property),
16737 (gst_gio_stream_src_get_property):
16738 * ext/gio/gstgiostreamsrc.h:
16739 Implement GstGioStreamSink and GstGioStreamSrc that have a property
16740 to set the GInputStream/GOutputStream that should be used.
16741 * tests/check/Makefile.am:
16742 * tests/check/pipelines/.cvsignore:
16743 * tests/check/pipelines/gio.c: (message_handler), (GST_START_TEST),
16744 (gio_testsuite), (main):
16745 Add unit test for giostreamsrc and giostreamsink.
16747 2007-11-07 11:48:09 +0000 Sebastian Dröge <slomo@circular-chaos.org>
16749 ext/gio/gstgio.c: Remove nowadays unnecessary workaround for a crash.
16750 Original commit message from CVS:
16751 * ext/gio/gstgio.c: (plugin_init):
16752 Remove nowadays unnecessary workaround for a crash.
16753 * ext/gio/gstgiosink.c: (gst_gio_sink_finalize),
16754 (gst_gio_sink_start), (gst_gio_sink_stop),
16755 (gst_gio_sink_unlock_stop):
16756 * ext/gio/gstgiosink.h:
16757 * ext/gio/gstgiosrc.c: (gst_gio_src_finalize), (gst_gio_src_start),
16758 (gst_gio_src_stop), (gst_gio_src_unlock_stop):
16759 * ext/gio/gstgiosrc.h:
16760 Make the finalize function safer, clean up everything that could stay
16762 Reset the cancellable instead of creating a new one after cancelling
16764 Don't store the GFile in the element, it's only necessary for creating
16767 2007-11-06 23:35:39 +0000 Sebastien Moutte <sebastien@moutte.net>
16769 gst-libs/gst/rtp/: Fix some C99-isms and and a missing function that some versions of
16770 Original commit message from CVS:
16771 Patch by: Sebastien Moutte <sebastien moutte net>
16772 * gst-libs/gst/rtp/gstrtcpbuffer.c: (gst_rtcp_ntp_to_unix),
16773 (gst_rtcp_unix_to_ntp):
16774 * gst-libs/gst/rtp/gstrtppayloads.c: (gst_rtp_payload_info_for_name):
16775 Fix some C99-isms and and a missing function that some versions of
16776 MSVC don't like too much (#494346).
16777 * win32/vs6/gst_plugins_base.dsw:
16778 * win32/vs6/libgstaudio.dsp:
16779 * win32/vs6/libgstrtp.dsp:
16780 * win32/vs6/libgsttag.dsp:
16781 Update vs6 projects files (#494346).
16783 2007-11-06 16:38:49 +0000 Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>
16785 win32/common/: More missing symbols to export (fixes #493986).
16786 Original commit message from CVS:
16787 Patch by: Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>
16788 * win32/common/libgstaudio.def:
16789 * win32/common/libgstcdda.def:
16790 * win32/common/libgstinterfaces.def:
16791 * win32/common/libgstnetbuffer.def:
16792 * win32/common/libgstpbutils.def:
16793 * win32/common/libgstrtp.def:
16794 * win32/common/libgstrtsp.def:
16795 * win32/common/libgsttag.def:
16796 * win32/common/libgstvideo.def:
16797 More missing symbols to export (fixes #493986).
16799 2007-11-06 11:58:59 +0000 Sebastian Dröge <slomo@circular-chaos.org>
16801 Remove the magnitude and phase calculation functions as these have very special use cases and can't even be used for ...
16802 Original commit message from CVS:
16803 * docs/libs/gst-plugins-base-libs-sections.txt:
16804 * gst-libs/gst/fft/gstfftf32.c:
16805 * gst-libs/gst/fft/gstfftf32.h:
16806 * gst-libs/gst/fft/gstfftf64.c:
16807 * gst-libs/gst/fft/gstfftf64.h:
16808 * gst-libs/gst/fft/gstffts16.c:
16809 * gst-libs/gst/fft/gstffts16.h:
16810 * gst-libs/gst/fft/gstffts32.c:
16811 * gst-libs/gst/fft/gstffts32.h:
16812 * tests/check/libs/fft.c: (GST_START_TEST):
16813 Remove the magnitude and phase calculation functions as these have
16814 very special use cases and can't even be used for the spectrum
16815 element. Also adjust the docs to mention some properties of the used
16816 FFT implemention, i.e. how the values are scaled. Fixes #492098.
16818 2007-11-06 11:09:30 +0000 Tim-Philipp Müller <tim@centricular.net>
16820 gst/playback/gstplaybasebin.c: Avoid crash when there are external subtitles (fixes #491722).
16821 Original commit message from CVS:
16822 * gst/playback/gstplaybasebin.c: (queue_threshold_reached),
16824 Avoid crash when there are external subtitles (fixes #491722).
16826 2007-11-03 10:39:21 +0000 Tim-Philipp Müller <tim@centricular.net>
16828 ext/alsa/: 'Could not open resource for writing' is not an acceptable even less so when we're trying to open it to re...
16829 Original commit message from CVS:
16830 * ext/alsa/gstalsasink.c: (gst_alsasink_open):
16831 * ext/alsa/gstalsasrc.c: (gst_alsasrc_open):
16832 'Could not open resource for writing' is not an acceptable
16833 error message when we can't open the audio device (see #492334),
16834 even less so when we're trying to open it to record something.
16836 2007-11-02 21:03:01 +0000 Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>
16838 win32/common/libgstrtp.def: Add some more missing symbols (#492813).
16839 Original commit message from CVS:
16840 Patch by: Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>
16841 * win32/common/libgstrtp.def:
16842 Add some more missing symbols (#492813).
16844 2007-11-02 14:59:06 +0000 Thijs Vermeir <thijsvermeir@gmail.com>
16846 tests/check/elements/audioconvert.c: Add check to make sure that the out caps have a channel layout set on them where...
16847 Original commit message from CVS:
16848 Patch by: Thijs Vermeir <thijsvermeir@gmail.com>
16849 * tests/check/elements/audioconvert.c: (verify_convert):
16850 Add check to make sure that the out caps have a channel layout
16851 set on them where they should have one.
16853 2007-11-01 13:28:59 +0000 Vincent Torri <vtorri@univ-evry.fr>
16855 gst-libs/gst/fft/: Include our own _stdint.h instead of sys/types.h, makes MingW happy (#492306).
16856 Original commit message from CVS:
16857 Patch by: Vincent Torri <vtorri at univ-evry dot fr>
16858 * gst-libs/gst/fft/kiss_fft_s16.h: (KISS_FFT_S16_MALLOC):
16859 * gst-libs/gst/fft/kiss_fft_s32.h: (KISS_FFT_S32_MALLOC):
16860 Include our own _stdint.h instead of sys/types.h, makes MingW happy
16862 * gst-libs/gst/rtsp/gstrtspconnection.c: (gst_rtsp_connection_create):
16863 Use _pipe directly, GLib doesn't have a pipe() macro any longer
16864 (it disappeared in GLib 2.14.0) (#492306).
16865 * gst-libs/gst/sdp/Makefile.am:
16866 * gst-libs/gst/sdp/gstsdpmessage.c:
16867 Fix includes and LIBS for win32/Mingw (#492306).
16868 * tests/examples/dynamic/addstream.c (pause_play_stream):
16869 Use more portable g_usleep() instead of sleep() (#492306).
16871 2007-11-01 12:51:57 +0000 Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>
16873 gst-libs/gst/audio/gstringbuffer.c: Return NULL instead of an enum that happens to be 0, fixes warning on MSVC (#4921...
16874 Original commit message from CVS:
16875 Patch by: Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>
16876 * gst-libs/gst/audio/gstringbuffer.c: (build_linear_format),
16877 (gst_ring_buffer_parse_caps):
16878 Return NULL instead of an enum that happens to be 0, fixes warning
16880 * gst-libs/gst/audio/gstringbuffer.h:
16881 No trailing commas in enum list (for gcc-2.9x).
16882 * gst/videotestsrc/videotestsrc.c: (random_char):
16883 Make information loss explicit instead of implicitly truncating to
16884 eight bits via the return value. Fixes runtime error on MSVC when
16885 using the debug CRT (#492114).
16886 * win32/common/config.h.in:
16887 Fix a bunch of '#undef FOO bar', which MSVC doesn't like (#492114).
16888 * win32/common/libgstinterfaces.def:
16889 * win32/common/libgstrtp.def:
16890 Export a few more symbols (#492114).
16892 2007-11-01 08:06:13 +0000 Stefan Kost <ensonic@users.sourceforge.net>
16894 gst-libs/gst/audio/audio.*: Readd the deprecation guards, but preserve compilability.
16895 Original commit message from CVS:
16896 * gst-libs/gst/audio/audio.c:
16897 * gst-libs/gst/audio/audio.h:
16898 Readd the deprecation guards, but preserve compilability.
16900 2007-10-31 17:54:48 +0000 Tim-Philipp Müller <tim@centricular.net>
16902 gst/audioconvert/gstaudioconvert.c: Preserve channel layout when fixating the number of channels in the output caps, ...
16903 Original commit message from CVS:
16904 * gst/audioconvert/gstaudioconvert.c: (find_suitable_channel_layout),
16905 (gst_audio_convert_fixate_channels), (gst_audio_convert_fixate_caps):
16906 Preserve channel layout when fixating the number of channels in the
16907 output caps, or make sure there's a suitable channel position layout
16908 set on the caps if required. Fixes #430677.
16910 2007-10-31 17:32:22 +0000 Tim-Philipp Müller <tim@centricular.net>
16912 tests/check/elements/decodebin.c: Make sure the pipeline really operates in push mode as it should in this case.
16913 Original commit message from CVS:
16914 * tests/check/elements/decodebin.c: (test_text_plain_streams):
16915 Make sure the pipeline really operates in push mode as it should
16918 2007-10-31 15:30:15 +0000 Tim-Philipp Müller <tim@centricular.net>
16920 gst-libs/gst/audio/audio.h: Don't guard deprecated enum with #ifndef DISABLE_DEPRECATED, or compilation with DISABLE_...
16921 Original commit message from CVS:
16922 * gst-libs/gst/audio/audio.h:
16923 Don't guard deprecated enum with #ifndef DISABLE_DEPRECATED, or
16924 compilation with DISABLE_DEPRECATED and without REMOVE_DEPRECATED
16925 (ie. normal cvs builds) will fail.
16927 2007-10-31 12:47:41 +0000 Stefan Kost <ensonic@users.sourceforge.net>
16929 tell gtk-doc about the deprecation guard. Apply more doc fixes.
16930 Original commit message from CVS:
16931 * docs/libs/Makefile.am:
16932 * gst-libs/gst/audio/audio.c:
16933 * gst-libs/gst/audio/audio.h:
16934 * gst-libs/gst/interfaces/mixer.c:
16935 tell gtk-doc about the deprecation guard. Apply more doc fixes.
16937 2007-10-31 12:30:28 +0000 Tim-Philipp Müller <tim@centricular.net>
16939 tests/check/libs/audio.c: Add simple unit test to make sure GstValue intersection of channel layouts works the way I ...
16940 Original commit message from CVS:
16941 * tests/check/libs/audio.c: (init_value_to_channel_layout),
16942 (test_channel_layout_value_intersect), (audio_suite):
16943 Add simple unit test to make sure GstValue intersection
16944 of channel layouts works the way I think it does.
16946 2007-10-30 20:32:14 +0000 Stefan Kost <ensonic@users.sourceforge.net>
16948 Fix the docs according to what gtk-doc complained about.
16949 Original commit message from CVS:
16950 * docs/libs/gst-plugins-base-libs-sections.txt:
16951 * gst-libs/gst/audio/gstaudiofilter.h:
16952 * gst-libs/gst/interfaces/mixer.h:
16953 * gst-libs/gst/rtp/gstbasertpdepayload.c:
16954 * gst-libs/gst/rtp/gstbasertpdepayload.h:
16955 * gst-libs/gst/sdp/gstsdpmessage.c:
16956 Fix the docs according to what gtk-doc complained about.
16958 2007-10-30 19:46:02 +0000 Stefan Kost <ensonic@users.sourceforge.net>
16960 tests/icles/stress-playbin.c: Fix the build.
16961 Original commit message from CVS:
16962 * tests/icles/stress-playbin.c:
16965 2007-10-30 15:54:46 +0000 Tim-Philipp Müller <tim@centricular.net>
16967 gst/playback/: Post nice/more useful error message if we don't have a decoder for the primary type.
16968 Original commit message from CVS:
16969 * gst/playback/gstdecodebin.c: (close_pad_link), (type_found):
16970 * gst/playback/gstdecodebin2.c: (analyze_new_pad):
16971 Post nice/more useful error message if we don't have a decoder for
16974 2007-10-30 15:07:58 +0000 Wim Taymans <wim.taymans@gmail.com>
16976 gst/playback/gstdecodebin2.c: Be a bit more useful, unblock the pads after we fired the no-more-pads signal so that w...
16977 Original commit message from CVS:
16978 * gst/playback/gstdecodebin2.c: (gst_decode_group_expose):
16979 Be a bit more useful, unblock the pads after we fired the no-more-pads
16980 signal so that we can use the signal to inspect and connect all pads
16981 without having to keep extra state outside of decodebin.
16983 2007-10-30 15:00:06 +0000 Wim Taymans <wim.taymans@gmail.com>
16985 gst/playback/gsturidecodebin.c: Implement default signal handler so that we return TRUE when nothing is connected.
16986 Original commit message from CVS:
16987 * gst/playback/gsturidecodebin.c:
16988 (gst_uri_decode_bin_autoplug_continue),
16989 (gst_uri_decode_bin_class_init), (no_more_pads_full):
16990 Implement default signal handler so that we return TRUE when nothing is
16993 2007-10-28 11:53:36 +0000 Sebastian Dröge <slomo@circular-chaos.org>
16995 gst-libs/gst/riff/riff-media.c: Use the ALSA channel layout as default for wav files without channel layout informati...
16996 Original commit message from CVS:
16997 * gst-libs/gst/riff/riff-media.c:
16998 (gst_riff_wavext_add_channel_layout),
16999 (gst_riff_wave_add_default_channel_layout),
17000 (gst_riff_wavext_get_default_channel_mask),
17001 (gst_riff_create_audio_caps):
17002 Use the ALSA channel layout as default for wav files without channel
17003 layout information. This fixes playback of chan-id.wav on 5.1 systems
17004 for example. Also refactor the channel layout setting a bit and add
17005 more default channel orders. Fixes #489010.
17007 2007-10-28 11:46:48 +0000 Sebastian Dröge <slomo@circular-chaos.org>
17010 Use the ALSA channel layout as default for wav files without channel layout information. This fixes playback of chan-...
17011 Original commit message from CVS:
17012 (gst_riff_wavext_add_channel_layout),
17013 (gst_riff_wave_add_default_channel_layout),
17014 (gst_riff_wavext_get_default_channel_mask),
17015 (gst_riff_create_audio_caps):
17016 Use the ALSA channel layout as default for wav files without channel
17017 layout information. This fixes playback of chan-id.wav on 5.1 systems
17018 for example. Also refactor the channel layout setting a bit and add
17019 more default channel orders. Fixes #489010.
17021 2007-10-26 18:57:33 +0000 Tim-Philipp Müller <tim@centricular.net>
17023 tests/check/libs/tag.c: GST_TAG_MUSICBRAINZ_SORTNAME is deprecated and we compile with
17024 Original commit message from CVS:
17025 * tests/check/libs/tag.c: (test_musicbrainz_tag_registration):
17026 GST_TAG_MUSICBRAINZ_SORTNAME is deprecated and we compile with
17027 -DGST_DISABLE_DEPRECATED, so use new GST_TAG_ARTIST_SORTNAME
17030 2007-10-26 12:07:14 +0000 Christian Schaller <uraeus@gnome.org>
17032 * gst-plugins-base.spec.in:
17034 Original commit message from CVS:
17037 2007-10-25 17:36:49 +0000 Wim Taymans <wim.taymans@gmail.com>
17039 gst/playback/gstdecodebin2.c: Move subtitle encoding property to decodebin2 so that it can set the property value on ...
17040 Original commit message from CVS:
17041 * gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init),
17042 (gst_decode_bin_dispose), (gst_decode_bin_set_caps),
17043 (gst_decode_bin_set_subs_encoding),
17044 (gst_decode_bin_get_subs_encoding), (gst_decode_bin_set_property),
17045 (gst_decode_bin_get_property), (analyze_new_pad):
17046 Move subtitle encoding property to decodebin2 so that it can set the
17047 property value on all elements that it autoplugs and that require it.
17048 Make caps refcounting more consistent in get/set.
17049 * gst/playback/gsturidecodebin.c: (_gst_boolean_accumulator),
17050 (gst_uri_decode_bin_class_init), (gst_uri_decode_bin_init),
17051 (gst_uri_decode_bin_finalize), (gst_uri_decode_bin_set_property),
17052 (gst_uri_decode_bin_get_property), (proxy_unknown_type_signal),
17053 (proxy_autoplug_continue_signal),
17054 (proxy_autoplug_factories_signal), (proxy_autoplug_select_signal),
17056 Proxy properties and relevant signals from the internal decodebin.
17057 Make properties MT safe.
17059 2007-10-25 15:10:59 +0000 Tim-Philipp Müller <tim@centricular.net>
17061 gst-libs/gst/tag/: Deprecate GST_TAG_MUSICBRAINZ_SORTNAME, replaced by the newly-added
17062 Original commit message from CVS:
17063 * gst-libs/gst/tag/tag.h: (GST_TAG_MUSICBRAINZ_SORTNAME):
17064 * gst-libs/gst/tag/tags.c:
17065 Deprecate GST_TAG_MUSICBRAINZ_SORTNAME, replaced by the newly-added
17066 GST_TAG_ARTIST_SORTNAME (in an API and ABI compatible way).
17067 * gst-libs/gst/tag/gstid3tag.c: (tag_matches):
17068 Map ID3v2 TSOP, TSOA and TSOT frames to new SORTNAME tags (#414539).
17069 * gst-libs/gst/tag/gstvorbistag.c: (tag_matches),
17070 (gst_tag_to_vorbis_comments):
17071 Map new SORTNAME tags (these tags aren't even semi-official, so I'm
17072 just mapping everything I found in the wild) (#414539).
17074 2007-10-24 11:07:57 +0000 Wim Taymans <wim.taymans@gmail.com>
17076 gst/playback/: Remove the autoplug-sort signal and replace it with a binding friendly autoplug-select signal.
17077 Original commit message from CVS:
17078 Inspired by patch of: René Stadler <mail at renestadler dot de>
17079 * gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init),
17080 (gst_decode_bin_autoplug_continue),
17081 (gst_decode_bin_autoplug_factories),
17082 (gst_decode_bin_autoplug_select), (analyze_new_pad), (connect_pad),
17083 (find_compatibles):
17084 * gst/playback/gstplay-marshal.list:
17085 Remove the autoplug-sort signal and replace it with a binding friendly
17086 autoplug-select signal.
17087 Add an autoplug-factories signal that can be used to generate a list of
17088 factories to try to autoplug.
17089 Add the GstPad to the autoplugging signal args as it might be needed to
17090 make a good factory selection.
17091 Fix up the marshallers for this. Fixes #407282.
17093 2007-10-23 14:23:14 +0000 Tim-Philipp Müller <tim@centricular.net>
17095 gst-libs/gst/tag/gsttagdemux.c: Don't abort with an assertion if we receive a seek event with a start type of NONE (s...
17096 Original commit message from CVS:
17097 * gst-libs/gst/tag/gsttagdemux.c:
17098 Don't abort with an assertion if we receive a seek event with
17099 a start type of NONE (see launchpad bug #155878).
17101 2007-10-22 10:21:46 +0000 Wim Taymans <wim.taymans@gmail.com>
17103 sys/: Make sure that before we clean up the X resources, we shutdown and join the event thread.
17104 Original commit message from CVS:
17105 * sys/ximage/ximagesink.c: (gst_ximagesink_event_thread),
17106 (gst_ximagesink_xcontext_get), (gst_ximagesink_xcontext_clear),
17107 (gst_ximagesink_change_state), (gst_ximagesink_reset):
17108 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_event_thread),
17109 (gst_xvimagesink_xcontext_get), (gst_xvimagesink_xcontext_clear),
17110 (gst_xvimagesink_change_state), (gst_xvimagesink_reset):
17111 Make sure that before we clean up the X resources, we shutdown and join
17113 Also make sure the event thread does not shut down immediatly after
17114 startup because the running variable is not yet correctly set.
17117 2007-10-16 16:48:38 +0000 Wim Taymans <wim.taymans@gmail.com>
17119 gst/playback/gstdecodebin.c: Make the window for a race in typefind and shutting down smaller until we figure out the...
17120 Original commit message from CVS:
17121 * gst/playback/gstdecodebin.c: (new_pad), (type_found):
17122 Make the window for a race in typefind and shutting down smaller until
17123 we figure out the right locking here. Avoids #485753 usually.
17124 * gst/playback/gstdecodebin2.c: (type_found), (pad_added_group_cb):
17125 Remove unneeded lock causing a race in typefind and shutting down.
17127 * gst/playback/gstplaybin.c: (gst_play_bin_change_state):
17128 Also remove sinks when going to NULL because we might not complete the
17129 state change to PAUSED, causing the PAUSED->READY state change not to
17132 2007-10-16 15:33:31 +0000 Wim Taymans <wim.taymans@gmail.com>
17134 gst-libs/gst/audio/gstbaseaudiosink.c: Also explicitly release the ringbuffer when going to NULL because it is requir...
17135 Original commit message from CVS:
17136 * gst-libs/gst/audio/gstbaseaudiosink.c:
17137 (gst_base_audio_sink_drain), (gst_base_audio_sink_change_state):
17138 Also explicitly release the ringbuffer when going to NULL because it
17139 is required in the setcaps function, before the state change to PAUSED
17142 2007-10-16 14:58:53 +0000 Tim-Philipp Müller <tim@centricular.net>
17144 tests/icles/: Does what it says on the tin.
17145 Original commit message from CVS:
17146 * tests/icles/.cvsignore:
17147 * tests/icles/Makefile.am:
17148 * tests/icles/stress-playbin.c:
17149 Does what it says on the tin.
17151 2007-10-15 11:38:39 +0000 Wim Taymans <wim.taymans@gmail.com>
17153 gst/playback/gstqueue2.c: Fix queue negotiation. See #486758.
17154 Original commit message from CVS:
17155 * gst/playback/gstqueue2.c: (gst_queue_init), (gst_queue_push_one):
17156 Fix queue negotiation. See #486758.
17158 2007-10-12 10:52:18 +0000 Jan Schmidt <thaytan@mad.scientist.com>
17160 Actual code change to go along with:
17161 Original commit message from CVS:
17162 Actual code change to go along with:
17163 2007-10-12 Jan Schmidt <Jan.Schmidt@sun.com>
17164 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xwindow_decorate),
17165 (gst_xvimagesink_xwindow_new),
17166 (gst_xvimagesink_update_colorbalance),
17167 (gst_xvimagesink_handle_xevents), (gst_xvimagesink_xcontext_get):
17168 Fix handling of some of the X atoms. If the last parameter is True,
17169 XInternAtom won't create the atom if it doesn't exist, and therefore
17170 might return None. This causes X errors on Xv implementations that
17171 don't provide the colour balance attributes.
17173 2007-10-12 10:37:09 +0000 Jan Schmidt <thaytan@mad.scientist.com>
17176 Remove stray character from the changelog.
17177 Original commit message from CVS:
17178 Remove stray character from the changelog.
17180 2007-10-12 10:33:27 +0000 Jan Schmidt <thaytan@mad.scientist.com>
17183 I'm too lazy to comment this
17184 Original commit message from CVS:
17185 *** empty log message ***
17187 2007-10-11 18:24:09 +0000 Tim-Philipp Müller <tim@centricular.net>
17189 Extract vorbis comment LICENSE tags correctly.
17190 Original commit message from CVS:
17191 * gst-libs/gst/tag/gstvorbistag.c:
17192 * tests/check/libs/tag.c:
17193 Extract vorbis comment LICENSE tags correctly.
17195 2007-10-11 16:12:21 +0000 Jason Kivlighn <jkivlighn@gmail.com>
17197 Map ID3v2 WCOP frame to GST_TAG_COPYRIGHT_URI (#447000).
17198 Original commit message from CVS:
17199 Patch by: Jason Kivlighn <jkivlighn gmail com>
17200 * gst-libs/gst/tag/gstid3tag.c:
17201 * tests/check/libs/tag.c:
17202 Map ID3v2 WCOP frame to GST_TAG_COPYRIGHT_URI (#447000).
17204 2007-10-10 17:01:51 +0000 Tim-Philipp Müller <tim@centricular.net>
17206 gst-libs/gst/tag/gsttagdemux.c: Don't error out when a buggy downstream element doesn't handle the newsegment event w...
17207 Original commit message from CVS:
17208 * gst-libs/gst/tag/gsttagdemux.c:
17209 Don't error out when a buggy downstream element doesn't
17210 handle the newsegment event we send properly (especially
17211 not without posting a meaningful error message on the
17212 bus). See bug #471370 and launchpad bug #136264.
17214 2007-10-10 15:36:56 +0000 Wim Taymans <wim.taymans@gmail.com>
17216 gst-libs/gst/audio/gstbaseaudiosink.c: Use new basesink method to make our EOS drain interruptable.
17217 Original commit message from CVS:
17218 * gst-libs/gst/audio/gstbaseaudiosink.c:
17219 (gst_base_audio_sink_drain):
17220 Use new basesink method to make our EOS drain interruptable.
17222 2007-10-10 09:37:09 +0000 Jan Schmidt <thaytan@mad.scientist.com>
17224 gst-libs/gst/rtp/gstrtppayloads.c: Fix silly search-replace oversight.
17225 Original commit message from CVS:
17226 * gst-libs/gst/rtp/gstrtppayloads.c:
17227 Fix silly search-replace oversight.
17229 2007-10-09 09:57:17 +0000 Laurent Glayal <spglegle@yahoo.fr>
17231 gst-libs/gst/rtp/gstbasertppayload.c: Fix caps memleak. Fixes #484989.
17232 Original commit message from CVS:
17233 Patch by: Laurent Glayal <spglegle at yahoo dot fr>
17234 * gst-libs/gst/rtp/gstbasertppayload.c: (copy_fixed),
17235 (gst_basertppayload_set_outcaps):
17236 Fix caps memleak. Fixes #484989.
17238 2007-10-08 18:04:34 +0000 Wim Taymans <wim.taymans@gmail.com>
17240 gst-libs/gst/rtp/gstbasertpdepayload.c: Fix debug output.
17241 Original commit message from CVS:
17242 * gst-libs/gst/rtp/gstbasertpdepayload.c:
17243 (gst_base_rtp_depayload_setcaps), (gst_base_rtp_depayload_chain):
17246 2007-10-08 18:02:53 +0000 Wim Taymans <wim.taymans@gmail.com>
17248 gst-libs/gst/audio/gstbaseaudiosrc.c: Also handle the case where there is no clock set on the audio source, like in t...
17249 Original commit message from CVS:
17250 * gst-libs/gst/audio/gstbaseaudiosrc.c:
17251 (gst_base_audio_src_create):
17252 Also handle the case where there is no clock set on the audio source,
17253 like in the unit tests.
17255 2007-10-08 17:40:17 +0000 Jan Schmidt <thaytan@mad.scientist.com>
17257 gst-libs/gst/rtp/gstrtppayloads.c: Use unsigned G_MAXUINT8 instead of -1 to initialise a guint8 to avoid compiler war...
17258 Original commit message from CVS:
17259 * gst-libs/gst/rtp/gstrtppayloads.c:
17260 Use unsigned G_MAXUINT8 instead of -1 to initialise a guint8
17261 to avoid compiler warnings
17263 2007-10-08 17:12:32 +0000 Wim Taymans <wim.taymans@gmail.com>
17265 gst/playback/: Don't disconnect the have_type signal because we never reconnect it later on. Instead keep a variable ...
17266 Original commit message from CVS:
17267 * gst/playback/gstdecodebin.c: (type_found),
17268 (gst_decode_bin_change_state):
17269 * gst/playback/gstdecodebin2.c: (type_found),
17270 (gst_decode_bin_change_state):
17271 Don't disconnect the have_type signal because we never reconnect it
17272 later on. Instead keep a variable to see if we already detected a type.
17274 2007-10-08 10:47:26 +0000 Wim Taymans <wim.taymans@gmail.com>
17276 gst/playback/: Unlink the signal handler when we found the type, we're not going to do anything sensible with more ty...
17277 Original commit message from CVS:
17278 * gst/playback/gstdecodebin.c: (add_fakesink), (type_found):
17279 * gst/playback/gstdecodebin2.c: (gst_decode_bin_init),
17281 Unlink the signal handler when we found the type, we're not going to do
17282 anything sensible with more type_found signals anyway.
17284 2007-10-08 06:07:22 +0000 Sebastian Dröge <slomo@circular-chaos.org>
17286 ext/gio/gstgio.c: Use GIO function to get a list of supported URI schemes instead of hard coding something.
17287 Original commit message from CVS:
17288 * ext/gio/gstgio.c: (gst_gio_get_supported_protocols):
17289 Use GIO function to get a list of supported URI schemes instead of
17290 hard coding something.
17292 2007-10-06 16:49:55 +0000 Tim-Philipp Müller <tim@centricular.net>
17294 gst-libs/gst/tag/gsttagdemux.c: Don't leak caps.
17295 Original commit message from CVS:
17296 * gst-libs/gst/tag/gsttagdemux.c:
17299 2007-10-06 15:04:53 +0000 Tim-Philipp Müller <tim@centricular.net>
17301 gst-libs/gst/tag/: API: add GstTagDemux base class for simple tag demuxers.
17302 Original commit message from CVS:
17303 * gst-libs/gst/tag/Makefile.am:
17304 * gst-libs/gst/tag/gsttagdemux.c:
17305 * gst-libs/gst/tag/gsttagdemux.h:
17306 API: add GstTagDemux base class for simple tag demuxers.
17307 * docs/libs/gst-plugins-base-libs-docs.sgml:
17308 * docs/libs/gst-plugins-base-libs-sections.txt:
17309 Add GstTagDemux to docs.
17311 2007-10-05 07:49:25 +0000 Sebastian Dröge <slomo@circular-chaos.org>
17313 gst-libs/gst/rtp/gstrtpbuffer.c: Fix bug introduced with last commit which inverted the logic and caused all buffers ...
17314 Original commit message from CVS:
17315 * gst-libs/gst/rtp/gstrtpbuffer.c:
17316 (gst_rtp_buffer_get_payload_subbuffer):
17317 Fix bug introduced with last commit which inverted the logic and
17318 caused all buffers to be dropped. Fixes #483620.
17319 Thanks to Laurent Glayal <spglegle at yahoo dot fr> for noticing.
17321 2007-10-04 06:50:53 +0000 Stefan Kost <ensonic@users.sourceforge.net>
17323 gst-libs/gst/rtp/gstrtpbuffer.c: with regular return and warning.
17324 Original commit message from CVS:
17325 * gst-libs/gst/rtp/gstrtpbuffer.c:
17326 Replace g_return_if_val (as it could be disabled), with regular return
17329 2007-10-03 14:51:59 +0000 Stefan Kost <ensonic@users.sourceforge.net>
17331 tests/check/pipelines/simple-launch-lines.c: Print message name and not just number.
17332 Original commit message from CVS:
17333 * tests/check/pipelines/simple-launch-lines.c:
17334 Print message name and not just number.
17336 2007-10-02 11:11:13 +0000 Wim Taymans <wim.taymans@gmail.com>
17338 gst-libs/gst/audio/gstbaseaudiosink.c: When slaved to the clock, don't try to align a sample with the previous one wh...
17339 Original commit message from CVS:
17340 * gst-libs/gst/audio/gstbaseaudiosink.c:
17341 (gst_base_audio_sink_async_play):
17342 When slaved to the clock, don't try to align a sample with the previous
17343 one when going to PLAYING again.
17345 2007-10-02 09:04:03 +0000 Stefan Kost <ensonic@users.sourceforge.net>
17347 tests/examples/snapshot/snapshot.c: Fix the build.
17348 Original commit message from CVS:
17349 * tests/examples/snapshot/snapshot.c:
17352 2007-10-02 07:43:57 +0000 Sebastian Dröge <slomo@circular-chaos.org>
17354 ext/gio/gstgiosink.c: Update to API changes in GIO.
17355 Original commit message from CVS:
17356 * ext/gio/gstgiosink.c: (gst_gio_sink_start):
17357 Update to API changes in GIO.
17359 2007-10-01 16:33:00 +0000 Wim Taymans <wim.taymans@gmail.com>
17361 gst-libs/gst/sdp/gstsdpmessage.h: Add RFC 3556 bandwidth modifiers.
17362 Original commit message from CVS:
17363 * gst-libs/gst/sdp/gstsdpmessage.h:
17364 Add RFC 3556 bandwidth modifiers.
17366 2007-10-01 13:37:31 +0000 Wim Taymans <wim.taymans@gmail.com>
17368 Update documentation.
17369 Original commit message from CVS:
17370 * docs/libs/gst-plugins-base-libs-docs.sgml:
17371 * docs/libs/gst-plugins-base-libs-sections.txt:
17372 * gst-libs/gst/rtp/gstrtppayloads.c:
17373 Update documentation.
17375 2007-10-01 13:22:14 +0000 Wim Taymans <wim.taymans@gmail.com>
17377 gst-libs/gst/rtp/: Added new file and header to deal with payload info.
17378 Original commit message from CVS:
17379 * gst-libs/gst/rtp/Makefile.am:
17380 * gst-libs/gst/rtp/gstrtppayloads.c: (gst_rtp_payload_info_for_pt),
17381 (gst_rtp_payload_info_for_name):
17382 * gst-libs/gst/rtp/gstrtppayloads.h:
17383 Added new file and header to deal with payload info.
17384 * gst-libs/gst/rtp/gstrtpbuffer.c: (gst_rtp_buffer_validate_data),
17385 (gst_rtp_buffer_default_clock_rate):
17386 * gst-libs/gst/rtp/gstrtpbuffer.h:
17387 Payload specific stuff is move to new headers.
17388 Implement _default_clock rate using the new payload function.
17389 * gst-libs/gst/sdp/gstsdpmessage.c: (is_multicast_address),
17390 (gst_sdp_parse_line):
17391 * gst-libs/gst/sdp/gstsdpmessage.h:
17392 Add some more comments.
17394 2007-10-01 10:22:46 +0000 Wim Taymans <wim.taymans@gmail.com>
17396 gst/typefind/gsttypefindfunctions.c: Add typefind function for application/sdp.
17397 Original commit message from CVS:
17398 * gst/typefind/gsttypefindfunctions.c: (utf8_type_find),
17399 (sdp_check_header), (sdp_type_find), (plugin_init):
17400 Add typefind function for application/sdp.
17401 Remove some old dirac typefind code that was ifdeffed out.
17403 2007-09-29 12:04:02 +0000 Sébastien Moutte <sebastien@moutte.net>
17405 win32/common/libgstaudio.def: Add new exported functions.
17406 Original commit message from CVS:
17407 * win32/common/libgstaudio.def:
17408 Add new exported functions.
17409 * win32/vs6/grammar.dsp:
17410 Add autogeneration and copy of some autegenerated files from win32/common
17412 * win32/vs6/libgstaudioconvert.dsp:
17413 Add gstaudioquantize.c to the build.
17414 * win32/vs6/libgstinterfaces.dsp:
17415 Add videoorientation.c to the build.
17416 * win32/vs6/libgstriff.dsp:
17417 Add libgsttag to the link libraries list.
17418 * win32/vs6/libgstvolume.dsp:
17419 Add liboil to the link.
17420 * win32/vs6/gst_plugins_base.dsw:
17421 * win32/vs6/libgstrtsp.dsp:
17422 * win32/common/libgstrtsp.def:
17423 Add files to build libgstrtsp library.
17425 2007-09-29 07:01:55 +0000 Sebastian Dröge <slomo@circular-chaos.org>
17427 ext/gio/: Some minor cleanup and allow setting the location only when the element is not playing or paused.
17428 Original commit message from CVS:
17429 * ext/gio/gstgiosink.c: (gst_gio_sink_base_init),
17430 (gst_gio_sink_set_property), (gst_gio_sink_render):
17431 * ext/gio/gstgiosrc.c: (gst_gio_src_base_init),
17432 (gst_gio_src_set_property):
17433 Some minor cleanup and allow setting the location only when the
17434 element is not playing or paused.
17436 2007-09-26 15:14:37 +0000 Wim Taymans <wim.taymans@gmail.com>
17438 tests/examples/snapshot/snapshot.c: Print error when pipeline failed to construct.
17439 Original commit message from CVS:
17440 * tests/examples/snapshot/snapshot.c: (main):
17441 Print error when pipeline failed to construct.
17443 2007-09-25 19:06:47 +0000 Tim-Philipp Müller <tim@centricular.net>
17445 Add mappings for the new GST_TAG_COMPOSER for vorbis comments and ID3v2 tags.
17446 Original commit message from CVS:
17448 * gst-libs/gst/tag/gstid3tag.c:
17449 * gst-libs/gst/tag/gstvorbistag.c:
17450 Add mappings for the new GST_TAG_COMPOSER for vorbis comments
17453 2007-09-25 11:54:09 +0000 Tim-Philipp Müller <tim@centricular.net>
17455 gst-libs/gst/floatcast/floatcast.h: Don't include config.h in an installed public header, this might break compilatio...
17456 Original commit message from CVS:
17457 * gst-libs/gst/floatcast/floatcast.h:
17458 Don't include config.h in an installed public header, this
17459 might break compilation of applications that don't have such
17460 a header and doesn't necessarily do what it's supposed to do
17461 anyway (ie. check for the lrint/lrintf defines) (#442065).
17462 Add docs for the various macros and document how this header
17463 has to be used (link against libm, etc.); add a few FIXMEs;
17464 include math.h for non-c99 code path. Based on patch by
17467 2007-09-25 07:50:59 +0000 Sebastian Dröge <slomo@circular-chaos.org>
17469 configure.ac: Use AG_GST_ARG_WITH_PLUGINS and AG_GST_ARG_ENABLE_EXTERNAL instead of duplicating these macros in confi...
17470 Original commit message from CVS:
17472 Use AG_GST_ARG_WITH_PLUGINS and AG_GST_ARG_ENABLE_EXTERNAL instead
17473 of duplicating these macros in configure.ac.
17475 2007-09-22 17:58:22 +0000 Thomas Vander Stichele <thomas@apestaart.org>
17477 po/: Updated translations to 0.10.14
17478 Original commit message from CVS:
17482 Updated translations to 0.10.14
17484 2007-09-22 17:57:08 +0000 Thomas Vander Stichele <thomas@apestaart.org>
17488 Original commit message from CVS:
17491 2007-09-22 17:56:28 +0000 Thomas Vander Stichele <thomas@apestaart.org>
17493 po/pl.po: Added Polish translation.
17494 Original commit message from CVS:
17495 translated by: Jakub Bogusz <qboosh@pld-linux.org>
17497 Added Polish translation.
17499 2007-09-22 17:55:37 +0000 Thomas Vander Stichele <thomas@apestaart.org>
17501 po/fi.po: Added Finnish translation.
17502 Original commit message from CVS:
17503 translated by: Ilkka Tuohela <hile@iki.fi>
17505 Added Finnish translation.
17507 2007-09-22 17:54:50 +0000 Thomas Vander Stichele <thomas@apestaart.org>
17509 po/es.po: Added Spanish translation.
17510 Original commit message from CVS:
17511 translated by: Jorge González González <aloriel@gmail.com>
17513 Added Spanish translation.
17515 2007-09-22 17:53:59 +0000 Thomas Vander Stichele <thomas@apestaart.org>
17517 po/da.po: Added Danish translation.
17518 Original commit message from CVS:
17519 translated by: Mogens Jaeger <mogens@jaeger.tf>
17521 Added Danish translation.
17523 2007-09-22 17:52:56 +0000 Thomas Vander Stichele <thomas@apestaart.org>
17525 po/zh_CN.po: Added Chinese (simplified) translation.
17526 Original commit message from CVS:
17527 translated by: Funda Wang <fundawang@linux.net.cn>
17529 Added Chinese (simplified) translation.
17531 2007-09-22 17:51:45 +0000 Thomas Vander Stichele <thomas@apestaart.org>
17533 po/bg.po: Added Bulgarian translation.
17534 Original commit message from CVS:
17535 translated by: Alexander Shopov <ash@contact.bg>
17537 Added Bulgarian translation.
17539 2007-09-21 18:00:24 +0000 Sebastian Dröge <slomo@circular-chaos.org>
17541 docs/plugins/gst-plugins-bad-plugins.hierarchy: Update hierarchy.
17542 Original commit message from CVS:
17543 * docs/plugins/gst-plugins-bad-plugins.hierarchy:
17545 * ext/gio/gstgiosink.h:
17546 * ext/gio/gstgiosrc.h:
17547 Mark private fields of the instance structs private.
17549 2007-09-21 17:31:05 +0000 Sebastian Dröge <slomo@circular-chaos.org>
17551 docs/plugins/: Add the GIO plugin to the docs and do a make update while doing that.
17552 Original commit message from CVS:
17553 * docs/plugins/Makefile.am:
17554 * docs/plugins/gst-plugins-bad-plugins-docs.sgml:
17555 * docs/plugins/gst-plugins-bad-plugins-sections.txt:
17556 * docs/plugins/gst-plugins-bad-plugins.args:
17557 * docs/plugins/gst-plugins-bad-plugins.signals:
17558 * docs/plugins/inspect/plugin-bz2.xml:
17559 * docs/plugins/inspect/plugin-cdxaparse.xml:
17560 * docs/plugins/inspect/plugin-dfbvideosink.xml:
17561 * docs/plugins/inspect/plugin-dtsdec.xml:
17562 * docs/plugins/inspect/plugin-equalizer.xml:
17563 * docs/plugins/inspect/plugin-faac.xml:
17564 * docs/plugins/inspect/plugin-faad.xml:
17565 * docs/plugins/inspect/plugin-filter.xml:
17566 * docs/plugins/inspect/plugin-freeze.xml:
17567 * docs/plugins/inspect/plugin-gio.xml:
17568 * docs/plugins/inspect/plugin-gsm.xml:
17569 * docs/plugins/inspect/plugin-gstrtpmanager.xml:
17570 * docs/plugins/inspect/plugin-h264parse.xml:
17571 * docs/plugins/inspect/plugin-modplug.xml:
17572 * docs/plugins/inspect/plugin-mpeg2enc.xml:
17573 * docs/plugins/inspect/plugin-musepack.xml:
17574 * docs/plugins/inspect/plugin-musicbrainz.xml:
17575 * docs/plugins/inspect/plugin-nsfdec.xml:
17576 * docs/plugins/inspect/plugin-replaygain.xml:
17577 * docs/plugins/inspect/plugin-soundtouch.xml:
17578 * docs/plugins/inspect/plugin-spcdec.xml:
17579 * docs/plugins/inspect/plugin-spectrum.xml:
17580 * docs/plugins/inspect/plugin-speed.xml:
17581 * docs/plugins/inspect/plugin-tta.xml:
17582 * docs/plugins/inspect/plugin-videosignal.xml:
17583 * docs/plugins/inspect/plugin-xingheader.xml:
17584 * docs/plugins/inspect/plugin-xvid.xml:
17585 Add the GIO plugin to the docs and do a make update
17587 * ext/gio/gstgiosrc.c: (gst_gio_src_start):
17588 Fix a small memleak.
17590 2007-09-21 17:07:56 +0000 René Stadler <mail@renestadler.de>
17592 Add a GIO/GVFS plugin with source and sink elements. This will only be enabled when --enable-experimental is given to...
17593 Original commit message from CVS:
17594 Patch by: René Stadler <mail at renestadler dot de>
17597 * ext/gio/Makefile.am:
17598 * ext/gio/gstgio.c: (gst_gio_error), (gst_gio_seek),
17599 (gst_gio_get_supported_protocols),
17600 (gst_gio_uri_handler_get_type_sink),
17601 (gst_gio_uri_handler_get_type_src),
17602 (gst_gio_uri_handler_get_protocols), (gst_gio_uri_handler_get_uri),
17603 (gst_gio_uri_handler_set_uri), (gst_gio_uri_handler_init),
17604 (gst_gio_uri_handler_do_init), (plugin_init):
17605 * ext/gio/gstgio.h:
17606 * ext/gio/gstgiosink.c: (gst_gio_sink_base_init),
17607 (gst_gio_sink_class_init), (gst_gio_sink_init),
17608 (gst_gio_sink_finalize), (gst_gio_sink_set_property),
17609 (gst_gio_sink_get_property), (gst_gio_sink_start),
17610 (gst_gio_sink_stop), (gst_gio_sink_unlock),
17611 (gst_gio_sink_unlock_stop), (gst_gio_sink_event),
17612 (gst_gio_sink_render), (gst_gio_sink_query):
17613 * ext/gio/gstgiosink.h:
17614 * ext/gio/gstgiosrc.c: (gst_gio_src_base_init),
17615 (gst_gio_src_class_init), (gst_gio_src_init),
17616 (gst_gio_src_finalize), (gst_gio_src_set_property),
17617 (gst_gio_src_get_property), (gst_gio_src_start),
17618 (gst_gio_src_stop), (gst_gio_src_get_size),
17619 (gst_gio_src_is_seekable), (gst_gio_src_unlock),
17620 (gst_gio_src_unlock_stop), (gst_gio_src_check_get_range),
17621 (gst_gio_src_create):
17622 * ext/gio/gstgiosrc.h:
17623 Add a GIO/GVFS plugin with source and sink elements. This will
17624 only be enabled when --enable-experimental is given to configure
17625 for now as the GIO API is not stable yet. Fixes #476916.
17627 2007-09-21 14:37:26 +0000 Wim Taymans <wim.taymans@gmail.com>
17629 gst/playback/gstqueue2.c: Fix compilation wrt printf arguments.
17630 Original commit message from CVS:
17631 * gst/playback/gstqueue2.c: (gst_queue_push_one):
17632 Fix compilation wrt printf arguments.
17634 2007-09-20 17:38:10 +0000 Wim Taymans <wim.taymans@gmail.com>
17636 examples/app/appsrc_ex.c: Fix compilation after changing the name of a method.
17637 Original commit message from CVS:
17638 * examples/app/appsrc_ex.c: (main):
17639 Fix compilation after changing the name of a method.
17641 2007-09-20 14:09:24 +0000 Wim Taymans <wim.taymans@gmail.com>
17643 Add simple snapshot example program using appsink.
17644 Original commit message from CVS:
17646 * tests/examples/Makefile.am:
17647 * tests/examples/snapshot/.cvsignore:
17648 * tests/examples/snapshot/Makefile.am:
17649 * tests/examples/snapshot/snapshot.c: (main):
17650 Add simple snapshot example program using appsink.
17652 2007-09-20 13:59:50 +0000 Wim Taymans <wim.taymans@gmail.com>
17654 gst-libs/gst/app/gstappsink.*: Add properties, signals and actions to access the element even without linking to the ...
17655 Original commit message from CVS:
17656 * gst-libs/gst/app/gstappsink.c: (gst_app_marshal_OBJECT__VOID),
17657 (gst_app_sink_class_init), (gst_app_sink_init),
17658 (gst_app_sink_dispose), (gst_app_sink_finalize),
17659 (gst_app_sink_set_property), (gst_app_sink_get_property),
17660 (gst_app_sink_flush_unlocked), (gst_app_sink_start),
17661 (gst_app_sink_event), (gst_app_sink_getcaps),
17662 (gst_app_sink_set_caps), (gst_app_sink_get_caps),
17663 (gst_app_sink_is_eos), (gst_app_sink_pull_preroll),
17664 (gst_app_sink_pull_buffer):
17665 * gst-libs/gst/app/gstappsink.h:
17666 Add properties, signals and actions to access the element even without
17667 linking to the library.
17668 Fix some method names and signatures.
17670 2007-09-20 10:37:02 +0000 Stefan Kost <ensonic@users.sourceforge.net>
17672 tests/check/generic/states.c: Improved state change unit test.
17673 Original commit message from CVS:
17674 * tests/check/generic/states.c:
17675 Improved state change unit test.
17677 2007-09-19 18:16:58 +0000 Stefan Kost <ensonic@users.sourceforge.net>
17679 Ignore registries in any format.
17680 Original commit message from CVS:
17681 * docs/plugins/.cvsignore:
17682 * tests/check/.cvsignore:
17683 Ignore registries in any format.
17685 2007-09-19 16:09:57 +0000 Wim Taymans <wim.taymans@gmail.com>
17687 gst-libs/gst/rtp/gstbasertpdepayload.c: Only copy timestamp on outgoing packets if the depayloader did not set one.
17688 Original commit message from CVS:
17689 * gst-libs/gst/rtp/gstbasertpdepayload.c:
17690 (gst_base_rtp_depayload_chain),
17691 (gst_base_rtp_depayload_set_gst_timestamp):
17692 Only copy timestamp on outgoing packets if the depayloader did not set
17694 Also copy duration on outgoing packets.
17696 2007-09-19 15:55:08 +0000 Wim Taymans <wim.taymans@gmail.com>
17698 gst-libs/gst/rtp/gstbasertppayload.c: Fix compilation because of missing %d in printf.
17699 Original commit message from CVS:
17700 * gst-libs/gst/rtp/gstbasertppayload.c: (copy_fixed),
17701 (gst_basertppayload_set_outcaps):
17702 Fix compilation because of missing %d in printf.
17703 When fixating caps, fixate what we can and throw away all remaining
17704 unfixed caps, subclasses should do something smart if they need to.
17706 2007-09-19 12:04:21 +0000 Stefan Kost <ensonic@users.sourceforge.net>
17708 ext/gnomevfs/gstgnomevfssrc.c: Improve debug logs a bit and be more verbose if things go wrong.
17709 Original commit message from CVS:
17710 * ext/gnomevfs/gstgnomevfssrc.c:
17711 Improve debug logs a bit and be more verbose if things go wrong.
17713 2007-09-17 17:24:55 +0000 Jan Schmidt <thaytan@mad.scientist.com>
17715 Fix a bunch of compile warnings shown with Forte.
17716 Original commit message from CVS:
17717 * ext/pango/gsttextoverlay.c: (gst_text_overlay_init),
17718 (gst_text_overlay_set_property):
17719 * ext/vorbis/vorbisdec.c: (vorbis_handle_data_packet):
17720 * gst-libs/gst/audio/gstbaseaudiosink.c:
17721 (gst_base_audio_sink_render):
17722 * gst-libs/gst/rtp/gstrtcpbuffer.c: (gst_rtcp_ntp_to_unix),
17723 (gst_rtcp_unix_to_ntp):
17724 * gst-libs/gst/rtsp/gstrtspmessage.c: (gst_rtsp_message_get_type):
17725 * gst/playback/gstqueue2.c:
17726 * tests/examples/seek/seek.c: (set_scale):
17727 Fix a bunch of compile warnings shown with Forte.
17728 * gst/audiorate/gstaudiorate.c:
17729 Always pull in config.h before including any system headers.
17731 2007-09-17 16:22:17 +0000 Wim Taymans <wim.taymans@gmail.com>
17733 gst/playback/gstqueue2.c: Also fix #476514 for queue2.
17734 Original commit message from CVS:
17735 * gst/playback/gstqueue2.c: (update_buffering),
17736 (gst_queue_locked_flush), (gst_queue_locked_enqueue),
17737 (gst_queue_handle_sink_event), (gst_queue_chain),
17738 (gst_queue_push_one), (gst_queue_sink_activate_push),
17739 (gst_queue_src_activate_push), (gst_queue_src_activate_pull):
17740 Also fix #476514 for queue2.
17742 2007-09-16 19:31:06 +0000 Wim Taymans <wim.taymans@gmail.com>
17744 gst-libs/gst/rtp/gstbasertpdepayload.c: Remove code to deal with RTP to GST time conversion, we now just copy the GST...
17745 Original commit message from CVS:
17746 * gst-libs/gst/rtp/gstbasertpdepayload.c:
17747 (gst_base_rtp_depayload_init), (gst_base_rtp_depayload_setcaps),
17748 (gst_base_rtp_depayload_chain),
17749 (gst_base_rtp_depayload_handle_sink_event),
17750 (gst_base_rtp_depayload_push_full),
17751 (gst_base_rtp_depayload_set_gst_timestamp),
17752 (gst_base_rtp_depayload_change_state):
17753 Remove code to deal with RTP to GST time conversion, we now just copy
17754 the GST timestamp we receive to the outgoing buffers.
17755 Handle segment and flushes correctly.
17756 * gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_push):
17757 When we have no valid input timestamp, use the previous rtp timestamp on
17758 the outgoing RTP packet instead of the RTP base time.
17760 2007-09-16 01:56:21 +0000 David Schleef <ds@schleef.org>
17762 ext/alsa/: Change alsa alloca's to malloc to fix warnings on gcc-4.2.
17763 Original commit message from CVS:
17764 * ext/alsa/gstalsa.c:
17765 * ext/alsa/gstalsadeviceprobe.c:
17766 * ext/alsa/gstalsamixer.c:
17767 * ext/alsa/gstalsasink.c:
17768 * ext/alsa/gstalsasrc.c:
17769 Change alsa alloca's to malloc to fix warnings on gcc-4.2.
17771 2007-09-15 18:41:27 +0000 Wim Taymans <wim.taymans@gmail.com>
17773 gst-libs/gst/rtp/gstbasertppayload.c: Add some debug info when negotiating caps.
17774 Original commit message from CVS:
17775 * gst-libs/gst/rtp/gstbasertppayload.c:
17776 (gst_basertppayload_set_outcaps), (gst_basertppayload_push):
17777 Add some debug info when negotiating caps.
17779 2007-09-15 00:29:11 +0000 Wim Taymans <wim.taymans@gmail.com>
17781 gst-libs/gst/rtp/gstrtpbuffer.c: A buffer with an empty payload is also a valid buffer.
17782 Original commit message from CVS:
17783 * gst-libs/gst/rtp/gstrtpbuffer.c: (gst_rtp_buffer_validate_data):
17784 A buffer with an empty payload is also a valid buffer.
17786 2007-09-14 20:52:00 +0000 Wim Taymans <wim.taymans@gmail.com>
17788 gst-libs/gst/rtp/gstbasertppayload.c: Make sure we start our RTP timestamp from the random base RTP timestamp even if...
17789 Original commit message from CVS:
17790 * gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_event),
17791 (gst_basertppayload_set_outcaps), (gst_basertppayload_push),
17792 (gst_basertppayload_change_state):
17793 Make sure we start our RTP timestamp from the random base RTP
17794 timestamp even if the buffer timestamp starts from some random value.
17796 2007-09-14 16:56:16 +0000 Wim Taymans <wim.taymans@gmail.com>
17798 Add simple exmple app to demonstrate starting and pausing live and non-live bins in a PLAYING pipeline.
17799 Original commit message from CVS:
17801 * tests/examples/Makefile.am:
17802 * tests/examples/dynamic/.cvsignore:
17803 * tests/examples/dynamic/Makefile.am:
17804 * tests/examples/dynamic/addstream.c: (create_stream),
17805 (pause_play_stream), (message_received), (eos_message_received),
17806 (perform_step), (main):
17807 Add simple exmple app to demonstrate starting and pausing live and
17808 non-live bins in a PLAYING pipeline.
17810 2007-09-14 10:42:00 +0000 Julien Moutte <julien@moutte.net>
17812 gst/typefind/gsttypefindfunctions.c: Add some typefind for QCP files (RFC #3625)
17813 Original commit message from CVS:
17814 2007-09-14 Julien MOUTTE <julien@moutte.net>
17815 * gst/typefind/gsttypefindfunctions.c: (plugin_init): Add some
17816 typefind for QCP files (RFC #3625)
17818 2007-09-13 22:52:09 +0000 Wim Taymans <wim.taymans@gmail.com>
17820 gst-libs/gst/audio/gstbaseaudiosink.c: Disable pull mode scheduling, we're not ready for it yet and it subtly breaks ...
17821 Original commit message from CVS:
17822 * gst-libs/gst/audio/gstbaseaudiosink.c:
17823 (gst_base_audio_sink_init):
17824 Disable pull mode scheduling, we're not ready for it yet and it subtly
17825 breaks a lot of things.
17827 2007-09-12 17:35:52 +0000 Tim-Philipp Müller <tim@centricular.net>
17829 tests/check/elements/libvisual.c: Test all libvisual plugins, not just the first one; this reproduces bug #450336 qui...
17830 Original commit message from CVS:
17831 * tests/check/elements/libvisual.c:
17832 Test all libvisual plugins, not just the first one; this reproduces
17833 bug #450336 quite easily. Looks like a problem with the 'jess'
17836 2007-09-12 17:15:12 +0000 Tim-Philipp Müller <tim@centricular.net>
17838 tests/check/: Add basic libvisual test case in an attempt to reproduce bug #450336.
17839 Original commit message from CVS:
17840 * tests/check/Makefile.am:
17841 * tests/check/elements/.cvsignore:
17842 * tests/check/elements/libvisual.c:
17843 Add basic libvisual test case in an attempt to reproduce bug #450336.
17844 Doesn't reproduce that bug, but some other crasher instead (invalid
17845 free), at least with make elements/libvisual.forever and the bumscope
17846 plugin on x86-64/gutsy. Leaving test disabled for now.
17848 2007-09-12 08:38:21 +0000 Peter Kjellerstedt <pkj@axis.com>
17850 gst/: Printf format fixes (#476128).
17851 Original commit message from CVS:
17852 Patch by: Peter Kjellerstedt <pkj at axis com>
17853 * gst-libs/gst/app/gstappsink.c:
17854 * gst/flv/gstflvdemux.c:
17855 * gst/flv/gstflvparse.c:
17856 * gst/interleave/deinterleave.c:
17857 * gst/switch/gstswitch.c:
17858 Printf format fixes (#476128).
17860 2007-09-11 19:07:57 +0000 Tommi Myöhänen <ext-tommi.myohanen@nokia.com>
17862 gst-libs/gst/rtsp/gstrtspconnection.c: Make sure we can not cancel in the middle of receiving a message.
17863 Original commit message from CVS:
17864 Patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com>
17865 * gst-libs/gst/rtsp/gstrtspconnection.c:
17866 (gst_rtsp_connection_read_internal), (gst_rtsp_connection_read),
17867 (read_body), (gst_rtsp_connection_receive):
17868 Make sure we can not cancel in the middle of receiving a message.
17871 2007-09-11 11:29:12 +0000 Josep Torra Valles <josep@fluendo.com>
17873 gst/playback/gstplaybasebin.c: Increase upper limit for audio queue a bit; fixes preroll problem with playbin and dec...
17874 Original commit message from CVS:
17875 Patch by: Josep Torra Valles <josep@fluendo.com>
17876 * gst/playback/gstplaybasebin.c:
17877 Increase upper limit for audio queue a bit; fixes preroll problem
17878 with playbin and decodebin2 when playing a quicktime trailer with
17879 multichannel audio via http (#464666).
17881 2007-09-10 22:10:54 +0000 Wim Taymans <wim.taymans@gmail.com>
17883 gst-libs/gst/audio/gstbaseaudiosrc.*: Allow othe clocks than the internal clock to be used for the pipeline.
17884 Original commit message from CVS:
17885 * gst-libs/gst/audio/gstbaseaudiosrc.c:
17886 (gst_base_audio_src_class_init), (gst_base_audio_src_init),
17887 (gst_base_audio_src_provide_clock),
17888 (gst_base_audio_src_set_property),
17889 (gst_base_audio_src_get_property), (gst_base_audio_src_create):
17890 * gst-libs/gst/audio/gstbaseaudiosrc.h:
17891 Allow othe clocks than the internal clock to be used for the pipeline.
17892 Add property to disable clock provide.
17893 API: GstBaseAudioSrc::provide-clock
17895 2007-09-10 12:05:34 +0000 Stefan Kost <ensonic@users.sourceforge.net>
17897 gst/playback/gstdecodebin2.c: Don't leak request pads. Fixes #475395.
17898 Original commit message from CVS:
17899 * gst/playback/gstdecodebin2.c:
17900 Don't leak request pads. Fixes #475395.
17902 2007-09-09 10:25:43 +0000 René Stadler <mail@renestadler.de>
17904 sys/: Correctly chain up finalize with the parent class to prevent memory leaks. Fixes #474880.
17905 Original commit message from CVS:
17906 Patch by: René Stadler <mail at renestadler dot de>
17907 * sys/ximage/ximagesink.c: (gst_ximage_buffer_finalize),
17908 (gst_ximage_buffer_class_init):
17909 * sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_destroy),
17910 (gst_xvimage_buffer_class_init):
17911 Correctly chain up finalize with the parent class to prevent
17912 memory leaks. Fixes #474880.
17914 2007-09-09 04:08:48 +0000 Sebastian Dröge <slomo@circular-chaos.org>
17916 Revert the latest change: floating point samples are allowed to have any value, not only values in the range [-1,1]. ...
17917 Original commit message from CVS:
17918 * gst/volume/gstvolume.c: (volume_choose_func):
17919 * tests/check/elements/volume.c: (GST_START_TEST):
17920 Revert the latest change: floating point samples are allowed to
17921 have any value, not only values in the range [-1,1]. Thanks to Andy
17922 Wingo for noticing.
17923 Also fix processing of int32 samples with volumes > 4 by making the
17924 unity value smaller which prevents overflows.
17926 2007-09-07 17:37:03 +0000 Tim-Philipp Müller <tim@centricular.net>
17928 Fix gst_rtp_buffer_set_csrc() and enable csrc-related unit test checks.
17929 Original commit message from CVS:
17930 * gst-libs/gst/rtp/gstrtpbuffer.c:
17931 * tests/check/libs/rtp.c:
17932 Fix gst_rtp_buffer_set_csrc() and enable csrc-related unit test checks.
17934 2007-09-07 16:46:05 +0000 Haakon Sporsheim <haakon.sporsheim@tandberg.com>
17936 gst-libs/gst/rtp/gstrtpbuffer.c: Fix up GstRTPHeader helper struct so that compilers will not under any circumstances...
17937 Original commit message from CVS:
17938 Based on patch by: Haakon Sporsheim <haakon.sporsheim at tandberg com>
17939 * gst-libs/gst/rtp/gstrtpbuffer.c:
17940 Fix up GstRTPHeader helper struct so that compilers will not under
17941 any circumstances add padding in between our fields, as currently
17942 happens with MSVC on win32, because that would lead to us sending
17943 out RTP payloads with broken RTP headers (#471194).
17944 Fix assertion guards for gst_rtp_buffer_get_csrc() and _set_csrc().
17945 * tests/check/Makefile.am:
17946 * tests/check/libs/.cvsignore:
17947 * tests/check/libs/rtp.c:
17948 Add some simple unit tests for GstRTPBuffer. Some are disabled
17949 because the code tested still needs fixing (set_csrc() does not work).
17951 2007-09-07 15:05:24 +0000 Christian Schaller <uraeus@gnome.org>
17953 * gst-plugins-base.spec.in:
17954 update spec file to include latest RTSP libraries and headers and more
17955 Original commit message from CVS:
17956 update spec file to include latest RTSP libraries and headers and more
17958 2007-09-07 12:41:01 +0000 Tim-Philipp Müller <tim@centricular.net>
17960 win32/: Add rtsp enumtypes (#474384) and update others.
17961 Original commit message from CVS:
17963 * win32/common/gstrtsp-enumtypes.c:
17964 * win32/common/gstrtsp-enumtypes.h:
17965 * win32/common/interfaces-enumtypes.c:
17966 * win32/common/interfaces-enumtypes.h:
17967 * win32/common/multichannel-enumtypes.c:
17968 Add rtsp enumtypes (#474384) and update others.
17970 2007-09-06 20:31:50 +0000 Stefan Kost <ensonic@users.sourceforge.net>
17972 configure.ac: Fix configure check for HAVE_LIBXML_HTML.
17973 Original commit message from CVS:
17975 Fix configure check for HAVE_LIBXML_HTML.
17977 2007-09-06 12:14:25 +0000 Tim-Philipp Müller <tim@centricular.net>
17979 tests/check/libs/.cvsignore: Ignore more, in case the build bots work again one day.
17980 Original commit message from CVS:
17981 * tests/check/libs/.cvsignore:
17982 Ignore more, in case the build bots work again one day.
17984 2007-09-06 07:00:36 +0000 Sebastian Dröge <slomo@circular-chaos.org>
17986 Add libgstfft, a FFT library based on Kiss FFT which is
17987 Original commit message from CVS:
17988 Reviewed by: Stefan Kost <ensonic@users.sf.net>
17990 * gst-libs/gst/Makefile.am:
17991 * gst-libs/gst/fft/Makefile.am:
17992 * gst-libs/gst/fft/_kiss_fft_guts_f32.h:
17993 * gst-libs/gst/fft/_kiss_fft_guts_f64.h:
17994 * gst-libs/gst/fft/_kiss_fft_guts_s16.h:
17995 * gst-libs/gst/fft/_kiss_fft_guts_s32.h:
17996 * gst-libs/gst/fft/gstfft.c: (gst_fft_next_fast_length):
17997 * gst-libs/gst/fft/gstfft.h:
17998 * gst-libs/gst/fft/gstfftf32.c: (gst_fft_f32_new),
17999 (gst_fft_f32_fft), (gst_fft_f32_inverse_fft), (gst_fft_f32_free),
18000 (gst_fft_f32_window), (gst_fft_f32_magnitude), (gst_fft_f32_phase):
18001 * gst-libs/gst/fft/gstfftf32.h:
18002 * gst-libs/gst/fft/gstfftf64.c: (gst_fft_f64_new),
18003 (gst_fft_f64_fft), (gst_fft_f64_inverse_fft), (gst_fft_f64_free),
18004 (gst_fft_f64_window), (gst_fft_f64_magnitude), (gst_fft_f64_phase):
18005 * gst-libs/gst/fft/gstfftf64.h:
18006 * gst-libs/gst/fft/gstffts16.c: (gst_fft_s16_new),
18007 (gst_fft_s16_fft), (gst_fft_s16_inverse_fft), (gst_fft_s16_free),
18008 (gst_fft_s16_window), (gst_fft_s16_magnitude), (gst_fft_s16_phase):
18009 * gst-libs/gst/fft/gstffts16.h:
18010 * gst-libs/gst/fft/gstffts32.c: (gst_fft_s32_new),
18011 (gst_fft_s32_fft), (gst_fft_s32_inverse_fft), (gst_fft_s32_free),
18012 (gst_fft_s32_window), (gst_fft_s32_magnitude), (gst_fft_s32_phase):
18013 * gst-libs/gst/fft/gstffts32.h:
18014 * gst-libs/gst/fft/kiss_fft_f32.c: (kf_bfly2), (kf_bfly4),
18015 (kf_bfly3), (kf_bfly5), (kf_bfly_generic), (kf_work), (kf_factor),
18016 (kiss_fft_f32_alloc), (kiss_fft_f32_stride), (kiss_fft_f32),
18017 (kiss_fft_f32_cleanup), (kiss_fft_f32_next_fast_size):
18018 * gst-libs/gst/fft/kiss_fft_f32.h:
18019 * gst-libs/gst/fft/kiss_fft_f64.c: (kf_bfly2), (kf_bfly4),
18020 (kf_bfly3), (kf_bfly5), (kf_bfly_generic), (kf_work), (kf_factor),
18021 (kiss_fft_f64_alloc), (kiss_fft_f64_stride), (kiss_fft_f64),
18022 (kiss_fft_f64_cleanup), (kiss_fft_f64_next_fast_size):
18023 * gst-libs/gst/fft/kiss_fft_f64.h:
18024 * gst-libs/gst/fft/kiss_fft_s16.c: (kf_bfly2), (kf_bfly4),
18025 (kf_bfly3), (kf_bfly5), (kf_bfly_generic), (kf_work), (kf_factor),
18026 (kiss_fft_s16_alloc), (kiss_fft_s16_stride), (kiss_fft_s16),
18027 (kiss_fft_s16_cleanup), (kiss_fft_s16_next_fast_size):
18028 * gst-libs/gst/fft/kiss_fft_s16.h:
18029 * gst-libs/gst/fft/kiss_fft_s32.c: (kf_bfly2), (kf_bfly4),
18030 (kf_bfly3), (kf_bfly5), (kf_bfly_generic), (kf_work), (kf_factor),
18031 (kiss_fft_s32_alloc), (kiss_fft_s32_stride), (kiss_fft_s32),
18032 (kiss_fft_s32_cleanup), (kiss_fft_s32_next_fast_size):
18033 * gst-libs/gst/fft/kiss_fft_s32.h:
18034 * gst-libs/gst/fft/kiss_fftr_f32.c: (kiss_fftr_f32_alloc),
18035 (kiss_fftr_f32), (kiss_fftri_f32):
18036 * gst-libs/gst/fft/kiss_fftr_f32.h:
18037 * gst-libs/gst/fft/kiss_fftr_f64.c: (kiss_fftr_f64_alloc),
18038 (kiss_fftr_f64), (kiss_fftri_f64):
18039 * gst-libs/gst/fft/kiss_fftr_f64.h:
18040 * gst-libs/gst/fft/kiss_fftr_s16.c: (kiss_fftr_s16_alloc),
18041 (kiss_fftr_s16), (kiss_fftri_s16):
18042 * gst-libs/gst/fft/kiss_fftr_s16.h:
18043 * gst-libs/gst/fft/kiss_fftr_s32.c: (kiss_fftr_s32_alloc),
18044 (kiss_fftr_s32), (kiss_fftri_s32):
18045 * gst-libs/gst/fft/kiss_fftr_s32.h:
18046 * gst-libs/gst/fft/kiss_version:
18047 * pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
18048 * pkgconfig/gstreamer-plugins-base.pc.in:
18049 Add libgstfft, a FFT library based on Kiss FFT which is
18050 BSD licensed. Supported sample formats are int16, int32,
18051 float and double. For those formats a real FFT and IFFT
18052 can be done, different windowing functions can be applied
18053 and functions for extracting the magnitude and phase exist.
18055 * docs/libs/Makefile.am:
18056 * docs/libs/gst-plugins-base-libs-docs.sgml:
18057 * docs/libs/gst-plugins-base-libs-sections.txt:
18058 Integrate libgstfft into the docs.
18059 * tests/check/Makefile.am:
18060 * tests/check/libs/fft.c: (GST_START_TEST), (fft_suite), (main):
18061 Add unit tests for libgstfft, currently only testing the FFT.
18062 Unit tests for IFFT will follow soon.
18064 2007-09-05 23:07:40 +0000 Peter Kjellerstedt <pkj@axis.com>
18066 gst-libs/gst/sdp/gstsdpmessage.*: Separate INIT_ARRAY() and related macros into two versions, one for structures and ...
18067 Original commit message from CVS:
18068 Patch by: Peter Kjellerstedt <pkj at axis com>
18069 * gst-libs/gst/sdp/gstsdpmessage.c: (gst_sdp_time_init),
18070 (gst_sdp_message_init), (gst_sdp_message_uninit),
18071 (is_multicast_address), (gst_sdp_message_as_text),
18072 (gst_sdp_message_get_origin), (gst_sdp_message_set_connection),
18073 (gst_sdp_message_get_connection), (gst_sdp_message_add_bandwidth),
18074 (gst_sdp_message_add_time), (gst_sdp_message_add_zone),
18075 (gst_sdp_message_get_key), (gst_sdp_message_get_attribute_val_n),
18076 (gst_sdp_message_get_attribute_val), (gst_sdp_message_add_media),
18077 (gst_sdp_media_init), (gst_sdp_media_uninit),
18078 (gst_sdp_media_as_text), (gst_sdp_media_set_port_info),
18079 (gst_sdp_media_connections_len), (gst_sdp_media_add_connection),
18080 (gst_sdp_media_bandwidths_len), (gst_sdp_media_get_bandwidth),
18081 (gst_sdp_media_add_bandwidth), (gst_sdp_media_attributes_len),
18082 (gst_sdp_parse_line), (print_media), (gst_sdp_message_dump):
18083 * gst-libs/gst/sdp/gstsdpmessage.h:
18084 Separate INIT_ARRAY() and related macros into two versions, one for
18085 structures and one for pointers (e.g., INIT_ARRAY() and
18086 INIT_PTR_ARRAY()). This fixes a segmentation error on freeing the
18087 lists of emails and phone numbers.
18088 Add missing const as appropriate.
18089 Change all gint to guint since they all actually represent unsigned
18091 Do not use time as a variable name as it shadows the global time().
18092 Add gst_sdp_message_as_text() and gst_sdp_media_as_text().
18093 Actually implement gst_sdp_message_add_time().
18094 Make gst_sdp_message_add_time() take repeat times as an argument.
18095 Store repeat times in GstSDPTime as a GArray rather than as gchar**.
18096 Corrected the definition of gst_sdp_media_get_bandwidth() (was
18097 misspelled as badwidth).
18098 gst-indented and a little clean up. Fixes #471067.
18100 2007-09-05 21:20:12 +0000 Sebastian Dröge <slomo@circular-chaos.org>
18102 gst/volume/gstvolume.c: Correctly clamp float/double samples in the [-1.0,1.0] range to prevent weird effects.
18103 Original commit message from CVS:
18104 * gst/volume/gstvolume.c: (volume_choose_func),
18105 (volume_process_double), (volume_process_double_clamp),
18106 (volume_process_float_clamp):
18107 Correctly clamp float/double samples in the [-1.0,1.0] range to
18108 prevent weird effects.
18109 * tests/check/elements/volume.c: (GST_START_TEST), (volume_suite):
18110 Add unit tests for all samples types that had none before.
18112 2007-09-05 14:09:15 +0000 Tim-Philipp Müller <tim@centricular.net>
18114 gst-libs/gst/rtp/gstrtpbuffer.c: Need to include stdlib.h for abs() here too.
18115 Original commit message from CVS:
18116 * gst-libs/gst/rtp/gstrtpbuffer.c:
18117 Need to include stdlib.h for abs() here too.
18119 2007-09-05 14:01:25 +0000 Tim-Philipp Müller <tim@centricular.net>
18121 gst/playback/gststreaminfo.c: Fix build.
18122 Original commit message from CVS:
18123 * gst/playback/gststreaminfo.c:
18126 2007-09-05 10:32:09 +0000 Stefan Kost <ensonic@users.sourceforge.net>
18128 gst/playback/gststreaminfo.c: Clean up some half-disabled code and comment.
18129 Original commit message from CVS:
18130 * gst/playback/gststreaminfo.c:
18131 Clean up some half-disabled code and comment.
18133 2007-09-04 16:18:48 +0000 Wim Taymans <wim.taymans@gmail.com>
18135 gst-libs/gst/rtp/gstbasertpaudiopayload.c: Return FALSE from the event handler to let the parent class handle the event.
18136 Original commit message from CVS:
18137 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
18138 (gst_base_rtp_payload_audio_handle_event):
18139 Return FALSE from the event handler to let the parent class handle the
18141 * gst-libs/gst/rtp/gstbasertpdepayload.c:
18142 (gst_base_rtp_depayload_chain), (gst_base_rtp_depayload_push_full):
18143 Mark outgoing buffers as DISCONT if the incomming buffer was DISCONT.
18144 * gst-libs/gst/rtp/gstbasertppayload.c:
18145 Bump the MTU to 1400.
18147 2007-09-04 01:50:55 +0000 Johan Dahlin <johan@gnome.org>
18149 gst/typefind/gsttypefindfunctions.c (plugin_init): Add an audio/x-nsf typefind function for the nsfdec element.
18150 Original commit message from CVS:
18151 2007-09-03 Johan Dahlin <jdahlin@async.com.br>
18152 * gst/typefind/gsttypefindfunctions.c (plugin_init):
18153 Add an audio/x-nsf typefind function for the nsfdec element.
18155 2007-09-03 20:46:38 +0000 Renato Filho <renato.filho@indt.org.br>
18157 gst/playback/gstplaybasebin.c: Included "myth://" on stream_uris list for enable buffering to mythtv files
18158 Original commit message from CVS:
18159 * gst/playback/gstplaybasebin.c:
18160 Included "myth://" on stream_uris list for enable buffering to mythtv files
18162 2007-09-03 19:31:11 +0000 Wim Taymans <wim.taymans@gmail.com>
18164 Fix parsing of RB blocks.
18165 Original commit message from CVS:
18166 * docs/libs/gst-plugins-base-libs-sections.txt:
18167 * gst-libs/gst/rtp/gstrtcpbuffer.c: (gst_rtcp_packet_get_rb),
18168 (gst_rtcp_packet_sdes_copy_entry), (gst_rtcp_ntp_to_unix),
18169 (gst_rtcp_unix_to_ntp):
18170 * gst-libs/gst/rtp/gstrtcpbuffer.h:
18171 Fix parsing of RB blocks.
18173 Added helper functions to convert to/from UNIX and NTP time.
18174 API: gst_rtcp_ntp_to_unix()
18175 API: gst_rtcp_unix_to_ntp()
18176 * gst-libs/gst/rtp/gstrtpbuffer.c: (gst_rtp_buffer_validate_data),
18177 (gst_rtp_buffer_get_header_len),
18178 (gst_rtp_buffer_get_extension_data),
18179 (gst_rtp_buffer_get_payload_subbuffer),
18180 (gst_rtp_buffer_get_payload_len), (gst_rtp_buffer_get_payload),
18181 (gst_rtp_buffer_ext_timestamp):
18182 * gst-libs/gst/rtp/gstrtpbuffer.h:
18183 Fix some more docs.
18184 Implement handling of packets with extensions.
18185 Fix padding check in _validate().
18186 Added function to get extension data.
18187 API: gst_rtp_buffer_get_header_len()
18188 API: gst_rtp_buffer_get_extension_data()
18190 2007-09-03 19:19:35 +0000 Wim Taymans <wim.taymans@gmail.com>
18192 gst-libs/gst/rtp/gstbasertpdepayload.c: Add some more docs for the queue-delay property and fix a typo in a comment.
18193 Original commit message from CVS:
18194 * gst-libs/gst/rtp/gstbasertpdepayload.c:
18195 (gst_base_rtp_depayload_class_init),
18196 (gst_base_rtp_depayload_set_gst_timestamp):
18197 Add some more docs for the queue-delay property and fix a typo in a
18199 * gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_push):
18202 2007-09-03 19:17:33 +0000 Wim Taymans <wim.taymans@gmail.com>
18204 gst-libs/gst/audio/gstbaseaudiosink.c: When skew slaving, try to hover around the middle of a segment so that we at m...
18205 Original commit message from CVS:
18206 * gst-libs/gst/audio/gstbaseaudiosink.c:
18207 (gst_base_audio_sink_skew_slaving), (gst_base_audio_sink_render),
18208 (gst_base_audio_sink_change_state):
18209 When skew slaving, try to hover around the middle of a segment so that
18210 we at most drift by half a segment.
18211 If we are aligning in the oposite direction of the clock skew, we don't
18214 2007-08-31 21:07:20 +0000 Wim Taymans <wim.taymans@gmail.com>
18216 gst-libs/gst/rtp/gstbasertpdepayload.c: Be less silly with the segment start, just apply the clock-base to the timest...
18217 Original commit message from CVS:
18218 * gst-libs/gst/rtp/gstbasertpdepayload.c:
18219 (gst_base_rtp_depayload_setcaps),
18220 (gst_base_rtp_depayload_set_gst_timestamp):
18221 Be less silly with the segment start, just apply the clock-base to the
18224 2007-08-31 15:58:30 +0000 Wim Taymans <wim.taymans@gmail.com>
18226 gst-libs/gst/rtp/gstbasertpdepayload.*: Deprecate the queue handling thread thing and remove the code.
18227 Original commit message from CVS:
18228 * gst-libs/gst/rtp/gstbasertpdepayload.c:
18229 (gst_base_rtp_depayload_class_init),
18230 (gst_base_rtp_depayload_finalize),
18231 (gst_base_rtp_depayload_setcaps), (gst_base_rtp_depayload_chain),
18232 (gst_base_rtp_depayload_handle_sink_event),
18233 (gst_base_rtp_depayload_set_gst_timestamp),
18234 (gst_base_rtp_depayload_change_state):
18235 * gst-libs/gst/rtp/gstbasertpdepayload.h:
18236 Deprecate the queue handling thread thing and remove the code.
18237 Use new method to calculate the extended timestamp.
18239 2007-08-31 15:21:13 +0000 Wim Taymans <wim.taymans@gmail.com>
18241 gst-libs/gst/rtp/gstrtcpbuffer.c: Use g_strndup which does exactly what we want.
18242 Original commit message from CVS:
18243 * gst-libs/gst/rtp/gstrtcpbuffer.c:
18244 (gst_rtcp_packet_sdes_copy_entry):
18245 Use g_strndup which does exactly what we want.
18246 * gst-libs/gst/rtp/gstrtpbuffer.c: (gst_rtp_buffer_compare_seqnum),
18247 (gst_rtp_buffer_ext_timestamp):
18248 * gst-libs/gst/rtp/gstrtpbuffer.h:
18249 Add helper function to compare seqnums.
18250 Add helper function to calculate extended timestamps.
18251 API: gst_rtp_buffer_compare_seqnum()
18252 API: gst_rtp_buffer_ext_timestamp()
18254 2007-08-30 21:59:23 +0000 Wim Taymans <wim.taymans@gmail.com>
18256 gst-libs/gst/rtp/gstrtcpbuffer.*: Fix and document SDES item data function.
18257 Original commit message from CVS:
18258 * gst-libs/gst/rtp/gstrtcpbuffer.c:
18259 (gst_rtcp_packet_sdes_get_entry),
18260 (gst_rtcp_packet_sdes_copy_entry):
18261 * gst-libs/gst/rtp/gstrtcpbuffer.h:
18262 Fix and document SDES item data function.
18263 Add new function that makes a proper copy of SDES item data.
18264 API: gst_rtcp_packet_sdes_copy_entry()
18266 2007-08-30 07:29:55 +0000 Stefan Kost <ensonic@users.sourceforge.net>
18268 The tcp and subparse plugins are under gst, but not totaly free of dependencies. Handle selection inconfigure.ac, so ...
18269 Original commit message from CVS:
18272 The tcp and subparse plugins are under gst, but not totaly free of
18273 dependencies. Handle selection inconfigure.ac, so that they show up
18274 on the final list of what is build and what is not. Maybe they should
18275 better be moved to ext.
18277 2007-08-30 06:58:46 +0000 Daniel Díaz <yosoy@danieldiaz.org>
18279 Check if libxml provides HTML parser which subparse needs.
18280 Original commit message from CVS:
18281 Patch by: Daniel Díaz <yosoy@danieldiaz.org>
18284 Check if libxml provides HTML parser which subparse needs.
18287 2007-08-29 14:22:04 +0000 Tim-Philipp Müller <tim@centricular.net>
18289 ext/alsa/gstalsa.c: Fix typo and compilation on big endian systems.
18290 Original commit message from CVS:
18291 * ext/alsa/gstalsa.c:
18292 Fix typo and compilation on big endian systems.
18294 2007-08-29 12:16:46 +0000 Tim-Philipp Müller <tim@centricular.net>
18296 gst/subparse/gstssaparse.c: Convert SSA newline codes into actual newline characters (#470766).
18297 Original commit message from CVS:
18298 * gst/subparse/gstssaparse.c:
18299 Convert SSA newline codes into actual newline characters (#470766).
18301 2007-08-28 14:58:17 +0000 Tim-Philipp Müller <tim@centricular.net>
18303 API: also add gst_install_plugins_supported() while we're at it (see #470456).
18304 Original commit message from CVS:
18305 * docs/libs/gst-plugins-base-libs-sections.txt:
18306 * gst-libs/gst/pbutils/install-plugins.c:
18307 * gst-libs/gst/pbutils/install-plugins.h:
18308 * tests/check/libs/pbutils.c:
18309 API: also add gst_install_plugins_supported() while we're at it
18312 2007-08-28 14:23:55 +0000 Tim-Philipp Müller <tim@centricular.net>
18314 API: add gst_missing_*_installer_detail_new() convenience API so that applications that know exactly what they're mis...
18315 Original commit message from CVS:
18316 * docs/libs/gst-plugins-base-libs-sections.txt:
18317 * gst-libs/gst/pbutils/missing-plugins.c:
18318 * gst-libs/gst/pbutils/missing-plugins.h:
18319 * tests/check/libs/pbutils.c:
18320 API: add gst_missing_*_installer_detail_new() convenience API so
18321 that applications that know exactly what they're missing can request
18322 installer detail strings for those items directly instead of having
18323 to first create a dummy missing-plugin message and then get the
18324 installer detail string from that. Fixes #470456.
18326 2007-08-27 11:59:56 +0000 Jan Schmidt <thaytan@mad.scientist.com>
18328 gst/playback/gstdecodebin.c: We need to set up delayed-linking whenever the caps are non-fixed, not just when there a...
18329 Original commit message from CVS:
18330 * gst/playback/gstdecodebin.c: (close_pad_link):
18331 We need to set up delayed-linking whenever the caps are non-fixed,
18332 not just when there are multiple types - use gst_pad_is_fixed()
18335 2007-08-26 14:14:33 +0000 Tim-Philipp Müller <tim@centricular.net>
18337 gst-libs/gst/pbutils/missing-plugins.c: Add missing separator in PID fallback case.
18338 Original commit message from CVS:
18339 * gst-libs/gst/pbutils/missing-plugins.c:
18340 (gst_missing_plugin_message_get_installer_detail):
18341 Add missing separator in PID fallback case.
18343 2007-08-24 15:28:33 +0000 Jan Schmidt <thaytan@mad.scientist.com>
18345 ext/alsa/Makefile.am: There is no GST_PLUGINS_BASE_LIBS defined.
18346 Original commit message from CVS:
18347 * ext/alsa/Makefile.am:
18348 There is no GST_PLUGINS_BASE_LIBS defined.
18349 * ext/alsa/gstalsa.c:
18350 * ext/alsa/gstalsasink.c: (gst_alsasink_delay):
18351 * ext/alsa/gstalsasrc.c: (gst_alsasrc_delay):
18352 Add support for ALSA 24-bit formats.
18353 snd_pcm_delay can return an error code, especially
18354 during XRUNS. In that case, the best we can do is assume
18356 * gst/audioconvert/Makefile.am:
18357 Add flags from -base before any more-remote dependencies.
18359 2007-08-23 20:45:45 +0000 Davyd <davyd@madeley.id.au>
18361 gst/volume/gstvolume.*: Add support for int32, int24 and int8 to the volume element.
18362 Original commit message from CVS:
18363 Based on a patch by: Davyd <davyd at madeley dot id dot au>
18364 * gst/volume/gstvolume.c: (volume_choose_func),
18365 (volume_update_real_volume), (gst_volume_set_volume),
18366 (gst_volume_init), (volume_process_int32),
18367 (volume_process_int32_clamp), (volume_process_int24),
18368 (volume_process_int24_clamp), (volume_process_int16),
18369 (volume_process_int16_clamp), (volume_process_int8),
18370 (volume_process_int8_clamp), (volume_update_volume), (plugin_init):
18371 * gst/volume/gstvolume.h:
18372 Add support for int32, int24 and int8 to the volume element.
18375 2007-08-23 12:37:42 +0000 Tim-Philipp Müller <tim@centricular.net>
18377 tests/examples/Makefile.am: Fix even more.
18378 Original commit message from CVS:
18379 * tests/examples/Makefile.am:
18382 2007-08-23 10:58:42 +0000 Stefan Kost <ensonic@users.sourceforge.net>
18384 Revert unwanted commit. many thanks to moap. I want a fix for https://thomas.apestaart.org/moap/trac/ticket/239
18385 Original commit message from CVS:
18387 * docs/libs/Makefile.am:
18388 * docs/libs/gst-plugins-base-libs-docs.sgml:
18389 * docs/libs/gst-plugins-base-libs-sections.txt:
18390 * ext/gnomevfs/gstgnomevfssrc.c:
18391 * ext/gnomevfs/gstgnomevfssrc.h:
18392 * gst-libs/gst/Makefile.am:
18393 * pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
18394 * pkgconfig/gstreamer-plugins-base.pc.in:
18395 * sys/v4l/v4lsrc_calls.c:
18396 * tests/examples/Makefile.am:
18397 * win32/common/config.h:
18398 Revert unwanted commit. many thanks to moap. I want a fix for
18399 https://thomas.apestaart.org/moap/trac/ticket/239
18401 2007-08-23 08:33:43 +0000 Stefan Kost <ensonic@users.sourceforge.net>
18403 Original commit message from CVS:
18404 reviewed by: <delete if not using a buddy>
18405 patch by: <delete if not someone else's patch>
18407 * docs/libs/Makefile.am:
18408 * docs/libs/gst-plugins-base-libs-docs.sgml:
18409 * docs/libs/gst-plugins-base-libs-sections.txt:
18410 * ext/gnomevfs/gstgnomevfssrc.c:
18411 * ext/gnomevfs/gstgnomevfssrc.h:
18412 * gst-libs/gst/Makefile.am:
18413 * gst-libs/gst/audio/gstaudiofilter.h:
18414 * gst/typefind/gsttypefindfunctions.c:
18415 * gst/volume/gstvolume.c:
18416 * pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
18417 * pkgconfig/gstreamer-plugins-base.pc.in:
18418 * sys/v4l/v4lsrc_calls.c:
18419 * tests/examples/Makefile.am:
18420 * win32/common/config.h:
18422 2007-08-22 15:29:04 +0000 Wim Taymans <wim.taymans@gmail.com>
18424 gst-libs/gst/audio/audio.c: Clarify the docs a little.
18425 Original commit message from CVS:
18426 * gst-libs/gst/audio/audio.c:
18427 Clarify the docs a little.
18429 2007-08-22 11:20:28 +0000 Stefan Kost <ensonic@users.sourceforge.net>
18431 gst/volume/gstvolume.c: Enable liboil for float and add more details about problems with int16.
18432 Original commit message from CVS:
18433 * gst/volume/gstvolume.c:
18434 Enable liboil for float and add more details about problems with
18437 2007-08-21 15:43:24 +0000 Wim Taymans <wim.taymans@gmail.com>
18439 sys/v4l/gstv4lsrc.c: Fix RGB24 masks as spotted by _ke (Daniel G. Siegel) on IRC.
18440 Original commit message from CVS:
18441 * sys/v4l/gstv4lsrc.c: (gst_v4lsrc_palette_to_caps):
18442 Fix RGB24 masks as spotted by _ke (Daniel G. Siegel) on IRC.
18444 2007-08-21 12:08:43 +0000 Wim Taymans <wim.taymans@gmail.com>
18446 ext/vorbis/vorbisdec.c: When calculating the first timestamp of the buffers, don't go below 0 and clip the samples be...
18447 Original commit message from CVS:
18448 * ext/vorbis/vorbisdec.c: (vorbis_dec_push_forward):
18449 When calculating the first timestamp of the buffers, don't go below 0
18450 and clip the samples because the offset was on the eos page.
18453 2007-08-21 11:42:39 +0000 Wim Taymans <wim.taymans@gmail.com>
18455 ext/ogg/gstoggdemux.c: Also submit the eos page when trying to find the first timestamp.
18456 Original commit message from CVS:
18457 * ext/ogg/gstoggdemux.c: (gst_ogg_demux_read_chain),
18458 (gst_ogg_demux_collect_chain_info):
18459 Also submit the eos page when trying to find the first timestamp.
18462 2007-08-17 15:24:43 +0000 Sebastian Dröge <slomo@circular-chaos.org>
18464 gst-libs/gst/audio/audio.h: Use gst_util_uint64_scale() instead of doing the math with double for GST_FRAMES_TO_CLOCK...
18465 Original commit message from CVS:
18466 * gst-libs/gst/audio/audio.h:
18467 Use gst_util_uint64_scale() instead of doing the math
18468 with double for GST_FRAMES_TO_CLOCK_TIME() and
18469 GST_CLOCK_TIME_TO_FRAMES(). For large timestamps this
18470 prevents rounding errors. Fixes #467667.
18472 2007-08-17 13:42:49 +0000 Wim Taymans <wim.taymans@gmail.com>
18474 gst-libs/gst/rtsp/gstrtspconnection.*: Small cleanups.
18475 Original commit message from CVS:
18476 * gst-libs/gst/rtsp/gstrtspconnection.c:
18477 (gst_rtsp_connection_connect), (gst_rtsp_connection_write),
18478 (gst_rtsp_connection_read), (gst_rtsp_connection_poll):
18479 * gst-libs/gst/rtsp/gstrtspconnection.h:
18481 On shutdown, don't read the control socket yet.
18482 Set timeout value correctly in all cases.
18483 Add function to check if the server accepts reads or writes.
18484 API: gst_rtsp_connection_poll()
18485 * gst-libs/gst/rtsp/gstrtspdefs.h:
18486 Fix compilation with -pedantic.
18487 Add enum for _poll.
18489 2007-08-16 17:11:48 +0000 Wim Taymans <wim.taymans@gmail.com>
18491 gst-libs/gst/app/gstappsink.c: Override the preroll vmethod instead of overriding the render method twice.
18492 Original commit message from CVS:
18493 * gst-libs/gst/app/gstappsink.c: (gst_app_sink_class_init):
18494 Override the preroll vmethod instead of overriding the render method
18497 2007-08-16 16:06:21 +0000 Olivier Crete <tester@tester.ca>
18499 gst-libs/gst/rtp/gstbasertppayload.*: Add getcaps vfunc to basertppayload. See #465146.
18500 Original commit message from CVS:
18501 Patch by: Olivier Crete <tester at tester ca>
18502 * gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_init),
18503 (gst_basertppayload_getcaps):
18504 * gst-libs/gst/rtp/gstbasertppayload.h:
18505 Add getcaps vfunc to basertppayload. See #465146.
18507 2007-08-16 11:20:56 +0000 Wim Taymans <wim.taymans@gmail.com>
18509 gst/playback/gstplaybasebin.c: Only post buffering messages when we are a stream.
18510 Original commit message from CVS:
18511 * gst/playback/gstplaybasebin.c: (queue_threshold_reached):
18512 Only post buffering messages when we are a stream.
18514 2007-08-15 17:05:45 +0000 Tim-Philipp Müller <tim@centricular.net>
18516 gst-libs/gst/pbutils/: Small docs fix and addition.
18517 Original commit message from CVS:
18518 * gst-libs/gst/pbutils/install-plugins.c:
18519 * gst-libs/gst/pbutils/missing-plugins.c:
18520 Small docs fix and addition.
18522 2007-08-14 17:47:34 +0000 Wim Taymans <wim.taymans@gmail.com>
18524 gst-libs/gst/app/gstappsink.c: Don't use new API.
18525 Original commit message from CVS:
18526 * gst-libs/gst/app/gstappsink.c: (gst_app_sink_flush_unlocked):
18529 2007-08-14 17:38:05 +0000 Wim Taymans <wim.taymans@gmail.com>
18531 gst-libs/gst/app/gstappsink.*: Make love to appsink.
18532 Original commit message from CVS:
18533 * gst-libs/gst/app/gstappsink.c: (gst_app_sink_base_init),
18534 (gst_app_sink_class_init), (gst_app_sink_dispose),
18535 (gst_app_sink_flush_unlocked), (gst_app_sink_start),
18536 (gst_app_sink_stop), (gst_app_sink_event), (gst_app_sink_preroll),
18537 (gst_app_sink_render), (gst_app_sink_get_caps),
18538 (gst_app_sink_set_caps), (gst_app_sink_end_of_stream),
18539 (gst_app_sink_pull_preroll), (gst_app_sink_pull_buffer):
18540 * gst-libs/gst/app/gstappsink.h:
18541 Make love to appsink.
18542 Make it support pulling of the preroll buffer.
18543 Add docs and debug statements.
18544 Fix some races wrt to EOS handling and stopping.
18546 Implement FLUSHING.
18547 API: gst_app_sink_pull_preroll()
18549 2007-08-13 15:37:29 +0000 Tim-Philipp Müller <tim@centricular.net>
18551 tests/icles/: Add a dumb little test for textoverlay alignments.
18552 Original commit message from CVS:
18553 * tests/icles/.cvsignore:
18554 * tests/icles/Makefile.am:
18555 * tests/icles/test-textoverlay.c:
18556 Add a dumb little test for textoverlay alignments.
18558 2007-08-13 15:26:54 +0000 Dan Williams <dcbw@redhat.com>
18560 ext/pango/gsttextoverlay.*: API: add "line-alignment" property (#459334). Add gtk-doc blurb for "silent" property so ...
18561 Original commit message from CVS:
18562 Patch by: Dan Williams <dcbw redhat com>
18563 * ext/pango/gsttextoverlay.c:
18564 * ext/pango/gsttextoverlay.h:
18565 API: add "line-alignment" property (#459334). Add gtk-doc blurb for
18566 "silent" property so there's a Since tag in the API reference.
18568 2007-08-13 11:21:00 +0000 Thomas Vander Stichele <thomas@apestaart.org>
18572 Original commit message from CVS:
18575 2007-08-12 16:30:36 +0000 Wim Taymans <wim.taymans@gmail.com>
18577 gst-libs/gst/rtp/gstbasertppayload.*: Improve caps negotiation so that downstream elements can confiure certain RTP p...
18578 Original commit message from CVS:
18579 * gst-libs/gst/rtp/gstbasertppayload.c:
18580 (gst_basertppayload_set_outcaps):
18581 * gst-libs/gst/rtp/gstbasertppayload.h:
18582 Improve caps negotiation so that downstream elements can confiure
18583 certain RTP properties by fixing them on the caps. See #465146.
18586 2007-08-11 12:39:51 +0000 Tim-Philipp Müller <tim@centricular.net>
18588 Mark as deprecated some macros which were presumably meant to be private API and accidentally exposed in the public h...
18589 Original commit message from CVS:
18590 * docs/libs/gst-plugins-base-libs-sections.txt:
18591 * gst-libs/gst/rtp/gstbasertpdepayload.c:
18592 * gst-libs/gst/rtp/gstbasertpdepayload.h:
18593 Mark as deprecated some macros which were presumably meant to be
18594 private API and accidentally exposed in the public header file.
18595 Also actually _init() lock (only works at the moment because the
18596 struct is zeroed out when created and the initial values in the
18597 mutex struct are zeroes too). (#459585)
18599 2007-08-10 17:35:52 +0000 Stefan Kost <ensonic@users.sourceforge.net>
18601 docs/libs/Makefile.am: Remove cruft and do some cleanups.
18602 Original commit message from CVS:
18603 * docs/libs/Makefile.am:
18604 Remove cruft and do some cleanups.
18605 * docs/libs/gst-plugins-base-libs-docs.sgml:
18606 Prepare for comming gtkdoc features (rebase against online docs).
18608 2007-08-10 13:55:44 +0000 Michael Smith <msmith@xiph.org>
18610 gst/audiorate/gstaudiorate.c: Debug output fixes.
18611 Original commit message from CVS:
18612 * gst/audiorate/gstaudiorate.c: (gst_audio_rate_chain):
18613 Debug output fixes.
18614 * tests/check/elements/audiorate.c: (do_perfect_stream_test),
18616 Change the number of buffers used; 500 is too many and leads to
18619 2007-08-10 10:08:05 +0000 Tim-Philipp Müller <tim@centricular.net>
18621 gst/: Printf format fixes (#465028).
18622 Original commit message from CVS:
18623 * gst/playback/gstqueue2.c:
18624 * gst/videorate/gstvideorate.c:
18625 Printf format fixes (#465028).
18627 2007-08-09 15:44:02 +0000 Michael Smith <msmith@xiph.org>
18629 gst/audiorate/gstaudiorate.c: If we have a large (> 1 second) discontinuity, push a series of smaller buffers rather ...
18630 Original commit message from CVS:
18631 * gst/audiorate/gstaudiorate.c: (gst_audio_rate_chain):
18632 If we have a large (> 1 second) discontinuity, push a series of
18633 smaller buffers rather than a single very large buffer. Avoids
18634 unreasonably large single buffer allocations when encountering a
18636 * tests/check/elements/audiorate.c: (GST_START_TEST),
18638 Add a test for this.
18640 2007-08-09 12:06:43 +0000 Josep Torra Valles <josep@fluendo.com>
18642 gst/playback/gstplaybasebin.c: Fixes: #465015
18643 Original commit message from CVS:
18644 * gst/playback/gstplaybasebin.c: (group_commit),
18645 (queue_remove_probe), (queue_threshold_reached):
18646 Patch by: Josep Torra Valles <josep@fluendo.com>
18648 Make sure we remove the check_queues buffer probe from the
18649 correct queue to avoid racily going back to "buffering 99%" when
18650 buffering is actually complete.
18651 Also, fix the spelling of Josep's surname in the ChangeLog.
18653 2007-08-09 11:37:28 +0000 Stefan Kost <ensonic@users.sourceforge.net>
18655 ext/ogg/gstoggmux.c: Do not leak oggmux instance.
18656 Original commit message from CVS:
18657 * ext/ogg/gstoggmux.c:
18658 Do not leak oggmux instance.
18659 * ext/vorbis/vorbisenc.c:
18662 2007-08-09 10:51:55 +0000 Thomas Vander Stichele <thomas@apestaart.org>
18664 po/: Updated translations.
18665 Original commit message from CVS:
18671 Updated translations.
18673 2007-08-08 16:07:21 +0000 Yang Hong <hongyang@redflag-linux.com>
18675 ext/pango/gsttextoverlay.*: Add 'silent' property to GstTimeOverlay. Fixes #462979
18676 Original commit message from CVS:
18677 patch by: Yang Hong <hongyang@redflag-linux.com>
18678 * ext/pango/gsttextoverlay.c:
18679 * ext/pango/gsttextoverlay.h:
18680 Add 'silent' property to GstTimeOverlay. Fixes #462979
18682 2007-08-08 15:05:22 +0000 Josep Torre Valles <josep@fluendo.com>
18684 Add connection-speed property. Fixes #464690.
18685 Original commit message from CVS:
18686 Patch by: Josep Torre Valles <josep@fluendo.com>
18687 * docs/plugins/gst-plugins-base-plugins.args:
18688 * gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_class_init),
18689 (gst_uri_decode_bin_init), (gst_uri_decode_bin_set_property),
18690 (gst_uri_decode_bin_get_property), (gen_source_element):
18691 Add connection-speed property. Fixes #464690.
18693 2007-08-07 15:13:46 +0000 Damien Lespiau <damien.lespiau@gmail.com>
18695 Fix compilation on windows. Fixes #464320.
18696 Original commit message from CVS:
18697 Patch by: Damien Lespiau <damien dot lespiau at gmail dot com>
18699 * gst-libs/gst/rtsp/Makefile.am:
18700 * gst-libs/gst/rtsp/gstrtspconnection.c:
18701 (gst_rtsp_connection_connect):
18702 Fix compilation on windows. Fixes #464320.
18704 2007-08-07 14:14:54 +0000 Josep Torre Valles <josep@fluendo.com>
18706 gst/playback/: Move connection-speed property from playbin to playbasebin so that we can also configure it in source ...
18707 Original commit message from CVS:
18708 Patch by: Josep Torre Valles <josep@fluendo.com>
18709 * gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init),
18710 (gst_play_base_bin_init), (queue_threshold_reached),
18711 (gen_source_element), (setup_substreams),
18712 (gst_play_base_bin_set_property), (gst_play_base_bin_get_property),
18713 (gst_play_base_bin_get_streaminfo_value_array):
18714 * gst/playback/gstplaybasebin.h:
18715 * gst/playback/gstplaybin.c: (gst_play_bin_class_init),
18716 (gst_play_bin_set_property), (gst_play_bin_get_property),
18717 (gst_play_bin_handle_redirect_message):
18718 Move connection-speed property from playbin to playbasebin so that we
18719 can also configure it in source elements that have the connection-speed
18720 property. Fixes #464028.
18721 Add some debug info here and there.
18723 2007-08-06 16:42:22 +0000 Sebastian Dröge <slomo@circular-chaos.org>
18725 gst/audiotestsrc/gstaudiotestsrc.c: Properly respond to conversion queries. Fixes #464079.
18726 Original commit message from CVS:
18727 * gst/audiotestsrc/gstaudiotestsrc.c: (gst_audio_test_src_query):
18728 Properly respond to conversion queries. Fixes #464079.
18730 2007-08-03 19:53:11 +0000 Sebastian Dröge <slomo@circular-chaos.org>
18732 gst/audiotestsrc/gstaudiotestsrc.*: Add float/double and int32 support to audiotestsrc. Fixes #460422.
18733 Original commit message from CVS:
18734 * gst/audiotestsrc/gstaudiotestsrc.c: (gst_audio_test_src_init),
18735 (gst_audio_test_src_src_fixate), (gst_audio_test_src_setcaps),
18736 (gst_audio_test_src_init_sine_table),
18737 (gst_audio_test_src_change_wave), (gst_audio_test_src_create):
18738 * gst/audiotestsrc/gstaudiotestsrc.h:
18739 Add float/double and int32 support to audiotestsrc. Fixes #460422.
18740 Also set the default volume to the default value specified in the
18743 2007-08-03 19:40:14 +0000 Jens Granseuer <jensgr@gmx.net>
18745 gst/audioconvert/gstaudioquantize.c: Fix C89 incompatibilities and spelling of explanations. Fixes #463215.
18746 Original commit message from CVS:
18747 Patch by: Jens Granseuer <jensgr at gmx dot net>
18748 * gst/audioconvert/gstaudioquantize.c:
18749 Fix C89 incompatibilities and spelling of explanations. Fixes #463215.
18751 2007-08-03 15:44:01 +0000 Wim Taymans <wim.taymans@gmail.com>
18753 gst-libs/gst/rtsp/gstrtsptransport.c: Add rdt manager for rdt transport.
18754 Original commit message from CVS:
18755 * gst-libs/gst/rtsp/gstrtsptransport.c: (gst_rtsp_transport_parse):
18756 Add rdt manager for rdt transport.
18757 Fix parsing of RDT transport.
18759 2007-08-03 14:43:15 +0000 Jan Schmidt <thaytan@mad.scientist.com>
18761 configure.ac: Back to CVS
18762 Original commit message from CVS:
18766 === release 0.10.14 ===
18768 2007-08-03 14:41:46 +0000 Jan Schmidt <thaytan@mad.scientist.com>
18774 * docs/plugins/gst-plugins-base-plugins.args:
18775 * docs/plugins/inspect/plugin-adder.xml:
18776 * docs/plugins/inspect/plugin-alsa.xml:
18777 * docs/plugins/inspect/plugin-audioconvert.xml:
18778 * docs/plugins/inspect/plugin-audiorate.xml:
18779 * docs/plugins/inspect/plugin-audioresample.xml:
18780 * docs/plugins/inspect/plugin-audiotestsrc.xml:
18781 * docs/plugins/inspect/plugin-cdparanoia.xml:
18782 * docs/plugins/inspect/plugin-decodebin.xml:
18783 * docs/plugins/inspect/plugin-decodebin2.xml:
18784 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
18785 * docs/plugins/inspect/plugin-gdp.xml:
18786 * docs/plugins/inspect/plugin-gnomevfs.xml:
18787 * docs/plugins/inspect/plugin-libvisual.xml:
18788 * docs/plugins/inspect/plugin-ogg.xml:
18789 * docs/plugins/inspect/plugin-pango.xml:
18790 * docs/plugins/inspect/plugin-playbin.xml:
18791 * docs/plugins/inspect/plugin-subparse.xml:
18792 * docs/plugins/inspect/plugin-tcp.xml:
18793 * docs/plugins/inspect/plugin-theora.xml:
18794 * docs/plugins/inspect/plugin-typefindfunctions.xml:
18795 * docs/plugins/inspect/plugin-video4linux.xml:
18796 * docs/plugins/inspect/plugin-videorate.xml:
18797 * docs/plugins/inspect/plugin-videoscale.xml:
18798 * docs/plugins/inspect/plugin-videotestsrc.xml:
18799 * docs/plugins/inspect/plugin-volume.xml:
18800 * docs/plugins/inspect/plugin-vorbis.xml:
18801 * docs/plugins/inspect/plugin-ximagesink.xml:
18802 * docs/plugins/inspect/plugin-xvimagesink.xml:
18803 * gst-plugins-base.doap:
18804 * win32/common/config.h:
18806 Original commit message from CVS:
18809 2007-08-03 14:24:08 +0000 Jan Schmidt <thaytan@mad.scientist.com>
18827 Original commit message from CVS:
18830 2007-07-27 17:37:19 +0000 Jan Schmidt <thaytan@mad.scientist.com>
18832 tests/check/libs/audio.c: Fix the test to reflect the behaviour of gst_audio_clip_buffer.
18833 Original commit message from CVS:
18834 * tests/check/libs/audio.c: (GST_START_TEST):
18835 Fix the test to reflect the behaviour of gst_audio_clip_buffer.
18837 2007-07-27 17:10:47 +0000 Jan Schmidt <thaytan@mad.scientist.com>
18839 gst-libs/gst/audio/audio.c: When clipping a buffer with no timestamp, assume it is within the segment without warnings.
18840 Original commit message from CVS:
18841 * gst-libs/gst/audio/audio.c:
18842 When clipping a buffer with no timestamp, assume it is
18843 within the segment without warnings.
18846 2007-07-27 11:16:23 +0000 Wim Taymans <wim.taymans@gmail.com>
18848 gst-libs/gst/rtsp/gstrtspextension.c: Fire the signal on the object, not the interface.
18849 Original commit message from CVS:
18850 * gst-libs/gst/rtsp/gstrtspextension.c: (gst_rtsp_extension_send):
18851 Fire the signal on the object, not the interface.
18853 2007-07-27 09:17:19 +0000 Jan Schmidt <thaytan@mad.scientist.com>
18855 gst-libs/gst/rtsp/.cvsignore: Ber. Don't include the full path, idiot.
18856 Original commit message from CVS:
18857 * gst-libs/gst/rtsp/.cvsignore:
18858 Ber. Don't include the full path, idiot.
18860 2007-07-27 08:29:29 +0000 Jan Schmidt <thaytan@mad.scientist.com>
18862 gst-libs/gst/rtsp/.cvsignore: Ignore generated files.
18863 Original commit message from CVS:
18864 * gst-libs/gst/rtsp/.cvsignore:
18865 Ignore generated files.
18867 2007-07-26 19:57:15 +0000 Jan Schmidt <thaytan@mad.scientist.com>
18869 gst-libs/gst/: Move the rtspextension.h interface into gstrtspextension.h as part of libgstrtsp instead of libgstinte...
18870 Original commit message from CVS:
18871 * gst-libs/gst/interfaces/Makefile.am:
18872 * gst-libs/gst/interfaces/interfaces-marshal.list:
18873 * gst-libs/gst/interfaces/rtspextension.c:
18874 * gst-libs/gst/interfaces/rtspextension.h:
18875 * gst-libs/gst/rtsp/Makefile.am:
18876 * gst-libs/gst/rtsp/gstrtsp.h:
18877 * gst-libs/gst/rtsp/gstrtspextension.c:
18878 (gst_rtsp_extension_get_type), (gst_rtsp_extension_iface_init),
18879 (gst_rtsp_extension_detect_server),
18880 (gst_rtsp_extension_before_send), (gst_rtsp_extension_after_send),
18881 (gst_rtsp_extension_parse_sdp), (gst_rtsp_extension_setup_media),
18882 (gst_rtsp_extension_configure_stream),
18883 (gst_rtsp_extension_get_transports),
18884 (gst_rtsp_extension_stream_select), (gst_rtsp_extension_send):
18885 * gst-libs/gst/rtsp/gstrtspextension.h:
18886 * gst-libs/gst/rtsp/rtsp-marshal.list:
18887 Move the rtspextension.h interface into gstrtspextension.h
18888 as part of libgstrtsp instead of libgstinterfaces, because it's
18889 only for use within plugins, not applications.
18890 Add stuff to do the enum & marshal generation needed in libgstrtsp now.
18891 Use the GST_TYPE_RTSP_RESULT enum type for the return value of the
18892 signal that the GstRTSPExtension interface emits, since G_TYPE_ENUM
18895 2007-07-26 15:48:01 +0000 Wim Taymans <wim.taymans@gmail.com>
18897 gst-libs/gst/interfaces/: Fix marshaller for the send signal.
18898 Original commit message from CVS:
18899 * gst-libs/gst/interfaces/Makefile.am:
18900 * gst-libs/gst/interfaces/interfaces-marshal.list:
18901 * gst-libs/gst/interfaces/rtspextension.c:
18902 (gst_rtsp_extension_iface_init),
18903 (gst_rtsp_extension_stream_select), (gst_rtsp_extension_send):
18904 * gst-libs/gst/interfaces/rtspextension.h:
18905 Fix marshaller for the send signal.
18906 Add URL to stream selection interface method.
18908 2007-07-26 15:35:43 +0000 Jan Schmidt <thaytan@mad.scientist.com>
18910 gst-libs/gst/riff/Makefile.am: Pull in our dependencies from -base before those from outside.
18911 Original commit message from CVS:
18912 * gst-libs/gst/riff/Makefile.am:
18913 Pull in our dependencies from -base before those from outside.
18915 2007-07-26 14:33:01 +0000 Wim Taymans <wim.taymans@gmail.com>
18917 API: gst_rtsp_base64_decode_ip()
18918 Original commit message from CVS:
18919 * docs/libs/gst-plugins-base-libs-sections.txt:
18920 * gst-libs/gst/rtsp/gstrtspbase64.c: (gst_rtsp_base64_decode_ip):
18921 * gst-libs/gst/rtsp/gstrtspbase64.h:
18922 API: gst_rtsp_base64_decode_ip()
18923 Added function to decode Base64 in-place.
18925 2007-07-26 14:08:01 +0000 Jan Schmidt <thaytan@mad.scientist.com>
18927 tests/check/libs/.cvsignore: Ignore the mixer test binary.
18928 Original commit message from CVS:
18929 * tests/check/libs/.cvsignore:
18930 Ignore the mixer test binary.
18932 2007-07-26 10:00:37 +0000 Jan Schmidt <thaytan@mad.scientist.com>
18934 ext/vorbis/vorbisdec.c: Gratuitous comment change to trigger a rebuild on the buildbots.
18935 Original commit message from CVS:
18936 * ext/vorbis/vorbisdec.c: (vorbis_dec_push_forward):
18937 Gratuitous comment change to trigger a rebuild on the buildbots.
18939 2007-07-25 18:20:36 +0000 Wim Taymans <wim.taymans@gmail.com>
18941 gst-libs/gst/sdp/gstsdpmessage.*: Constify args where we can.
18942 Original commit message from CVS:
18943 * gst-libs/gst/sdp/gstsdpmessage.c: (gst_sdp_media_get_media),
18944 (gst_sdp_media_get_port), (gst_sdp_media_get_num_ports),
18945 (gst_sdp_media_get_proto), (gst_sdp_media_formats_len),
18946 (gst_sdp_media_get_format), (gst_sdp_media_get_information),
18947 (gst_sdp_media_connections_len), (gst_sdp_media_get_connection),
18948 (gst_sdp_media_bandwidths_len), (gst_sdp_media_get_badwidth),
18949 (gst_sdp_media_get_key), (gst_sdp_media_attributes_len),
18950 (gst_sdp_media_get_attribute), (gst_sdp_media_get_attribute_val_n),
18951 (gst_sdp_media_get_attribute_val):
18952 * gst-libs/gst/sdp/gstsdpmessage.h:
18953 Constify args where we can.
18955 2007-07-25 18:18:49 +0000 Wim Taymans <wim.taymans@gmail.com>
18957 gst-libs/gst/interfaces/: Move interface for RTSP extensions from -good to here.
18958 Original commit message from CVS:
18959 * gst-libs/gst/interfaces/Makefile.am:
18960 * gst-libs/gst/interfaces/rtspextension.c:
18961 (gst_rtsp_extension_get_type), (gst_rtsp_extension_iface_init),
18962 (gst_rtsp_extension_detect_server),
18963 (gst_rtsp_extension_before_send), (gst_rtsp_extension_after_send),
18964 (gst_rtsp_extension_parse_sdp), (gst_rtsp_extension_setup_media),
18965 (gst_rtsp_extension_configure_stream),
18966 (gst_rtsp_extension_get_transports),
18967 (gst_rtsp_extension_stream_select), (gst_rtsp_extension_send):
18968 * gst-libs/gst/interfaces/rtspextension.h:
18969 Move interface for RTSP extensions from -good to here.
18970 Added helper methods to invoke interface methods.
18972 2007-07-25 11:22:30 +0000 Wim Taymans <wim.taymans@gmail.com>
18974 Fix some more RTSP docs.
18975 Original commit message from CVS:
18976 * docs/libs/gst-plugins-base-libs-sections.txt:
18977 * gst-libs/gst/rtsp/gstrtspdefs.h:
18978 * gst-libs/gst/rtsp/gstrtspmessage.c: (key_value_foreach),
18979 (gst_rtsp_message_get_type), (gst_rtsp_message_parse_request),
18980 (gst_rtsp_message_init_response),
18981 (gst_rtsp_message_parse_response), (gst_rtsp_message_new_data),
18982 (gst_rtsp_message_parse_data), (gst_rtsp_message_add_header),
18983 (gst_rtsp_message_remove_header), (gst_rtsp_message_get_header),
18984 (gst_rtsp_message_append_headers), (gst_rtsp_message_set_body),
18985 (gst_rtsp_message_get_body), (dump_key_value):
18986 * gst-libs/gst/rtsp/gstrtspmessage.h:
18987 * gst-libs/gst/rtsp/gstrtsprange.c: (parse_npt_time),
18988 (parse_npt_range), (parse_clock_range), (parse_smpte_range),
18989 (gst_rtsp_range_parse):
18990 * gst-libs/gst/rtsp/gstrtsprange.h:
18991 * gst-libs/gst/rtsp/gstrtsptransport.c:
18992 * gst-libs/gst/rtsp/gstrtspurl.c:
18993 Fix some more RTSP docs.
18994 Add some missing methods for dealing with messages.
18996 2007-07-24 19:19:33 +0000 Wim Taymans <wim.taymans@gmail.com>
18998 Added beginnings of RTSP documentation.
18999 Original commit message from CVS:
19000 * docs/libs/gst-plugins-base-libs-docs.sgml:
19001 * docs/libs/gst-plugins-base-libs-sections.txt:
19002 * gst-libs/gst/rtsp/gstrtspbase64.c: (gst_rtsp_base64_encode):
19003 * gst-libs/gst/rtsp/gstrtspbase64.h:
19004 * gst-libs/gst/rtsp/gstrtspconnection.c:
19005 (gst_rtsp_connection_connect), (add_auth_header),
19006 (gst_rtsp_connection_write), (gst_rtsp_connection_send),
19007 (read_body), (gst_rtsp_connection_receive),
19008 (gst_rtsp_connection_next_timeout),
19009 (gst_rtsp_connection_reset_timeout),
19010 (gst_rtsp_connection_set_auth):
19011 * gst-libs/gst/rtsp/gstrtspconnection.h:
19012 * gst-libs/gst/rtsp/gstrtspdefs.c: (rtsp_init_status):
19013 * gst-libs/gst/rtsp/gstrtspdefs.h:
19014 * gst-libs/gst/rtsp/gstrtspmessage.h:
19015 * gst-libs/gst/rtsp/gstrtsprange.c: (parse_npt_time),
19016 (parse_npt_range), (parse_clock_range), (parse_smpte_range),
19017 (gst_rtsp_range_parse):
19018 * gst-libs/gst/rtsp/gstrtspurl.h:
19019 Added beginnings of RTSP documentation.
19021 2007-07-24 17:37:03 +0000 Wim Taymans <wim.taymans@gmail.com>
19023 Document the SDP library.
19024 Original commit message from CVS:
19025 * docs/libs/Makefile.am:
19026 * docs/libs/gst-plugins-base-libs-docs.sgml:
19027 * docs/libs/gst-plugins-base-libs-sections.txt:
19028 * gst-libs/gst/sdp/gstsdp.h:
19029 * gst-libs/gst/sdp/gstsdpmessage.c: (gst_sdp_message_set_origin),
19030 (gst_sdp_message_set_connection), (gst_sdp_message_add_bandwidth),
19031 (gst_sdp_message_add_time), (gst_sdp_message_add_zone),
19032 (gst_sdp_message_set_key), (gst_sdp_message_get_attribute_val_n),
19033 (gst_sdp_message_get_attribute_val),
19034 (gst_sdp_message_add_attribute), (gst_sdp_media_new),
19035 (gst_sdp_media_init), (gst_sdp_media_uninit), (gst_sdp_media_free),
19036 (gst_sdp_media_get_media), (gst_sdp_media_set_media),
19037 (gst_sdp_media_get_port), (gst_sdp_media_get_num_ports),
19038 (gst_sdp_media_set_port_info), (gst_sdp_media_get_proto),
19039 (gst_sdp_media_set_proto), (gst_sdp_media_formats_len),
19040 (gst_sdp_media_get_format), (gst_sdp_media_add_format),
19041 (gst_sdp_media_get_information), (gst_sdp_media_set_information),
19042 (gst_sdp_media_connections_len), (gst_sdp_media_get_connection),
19043 (gst_sdp_media_add_connection), (gst_sdp_media_bandwidths_len),
19044 (gst_sdp_media_get_badwidth), (gst_sdp_media_add_bandwidth),
19045 (gst_sdp_media_set_key), (gst_sdp_media_get_key),
19046 (gst_sdp_media_attributes_len), (gst_sdp_media_add_attribute),
19047 (gst_sdp_media_get_attribute_val_n),
19048 (gst_sdp_media_get_attribute_val), (gst_sdp_message_parse_buffer),
19049 (print_media), (gst_sdp_message_dump):
19050 * gst-libs/gst/sdp/gstsdpmessage.h:
19051 Document the SDP library.
19052 Add some of the missing SDPMedia methods.
19054 2007-07-24 11:52:56 +0000 Wim Taymans <wim.taymans@gmail.com>
19056 Move SDP and RTSP from helper objects in -good to a reusable library.
19057 Original commit message from CVS:
19059 * gst-libs/gst/Makefile.am:
19060 * gst-libs/gst/rtsp/Makefile.am:
19061 * gst-libs/gst/rtsp/gstrtspbase64.c: (gst_rtsp_base64_encode):
19062 * gst-libs/gst/rtsp/gstrtspbase64.h:
19063 * gst-libs/gst/rtsp/gstrtspconnection.c: (inet_aton),
19064 (gst_rtsp_connection_create), (gst_rtsp_connection_connect),
19065 (add_auth_header), (add_date_header), (gst_rtsp_connection_write),
19066 (gst_rtsp_connection_send), (read_line), (read_string), (read_key),
19067 (parse_response_status), (parse_request_line), (parse_line),
19068 (gst_rtsp_connection_read), (read_body),
19069 (gst_rtsp_connection_receive), (gst_rtsp_connection_close),
19070 (gst_rtsp_connection_free), (gst_rtsp_connection_next_timeout),
19071 (gst_rtsp_connection_reset_timeout), (gst_rtsp_connection_flush),
19072 (gst_rtsp_connection_set_auth):
19073 * gst-libs/gst/rtsp/gstrtspconnection.h:
19074 * gst-libs/gst/rtsp/gstrtspdefs.c: (rtsp_init_status),
19075 (gst_rtsp_strresult), (gst_rtsp_method_as_text),
19076 (gst_rtsp_version_as_text), (gst_rtsp_header_as_text),
19077 (gst_rtsp_status_as_text), (gst_rtsp_find_header_field),
19078 (gst_rtsp_find_method):
19079 * gst-libs/gst/rtsp/gstrtspdefs.h:
19080 * gst-libs/gst/rtsp/gstrtspmessage.c: (key_value_foreach),
19081 (gst_rtsp_message_new), (gst_rtsp_message_init),
19082 (gst_rtsp_message_new_request), (gst_rtsp_message_init_request),
19083 (gst_rtsp_message_new_response), (gst_rtsp_message_init_response),
19084 (gst_rtsp_message_init_data), (gst_rtsp_message_unset),
19085 (gst_rtsp_message_free), (gst_rtsp_message_add_header),
19086 (gst_rtsp_message_remove_header), (gst_rtsp_message_get_header),
19087 (gst_rtsp_message_append_headers), (gst_rtsp_message_set_body),
19088 (gst_rtsp_message_take_body), (gst_rtsp_message_get_body),
19089 (gst_rtsp_message_steal_body), (dump_mem), (dump_key_value),
19090 (gst_rtsp_message_dump):
19091 * gst-libs/gst/rtsp/gstrtspmessage.h:
19092 * gst-libs/gst/rtsp/gstrtsprange.c: (parse_npt_time),
19093 (parse_npt_range), (parse_clock_range), (parse_smpte_range),
19094 (gst_rtsp_range_parse), (gst_rtsp_range_free):
19095 * gst-libs/gst/rtsp/gstrtsprange.h:
19096 * gst-libs/gst/rtsp/gstrtsptransport.c: (gst_rtsp_transport_new),
19097 (gst_rtsp_transport_init), (gst_rtsp_transport_get_mime),
19098 (gst_rtsp_transport_get_manager), (parse_mode), (parse_range),
19099 (range_as_text), (rtsp_transport_mode_as_text),
19100 (rtsp_transport_profile_as_text), (rtsp_transport_ltrans_as_text),
19101 (gst_rtsp_transport_parse), (gst_rtsp_transport_as_text),
19102 (gst_rtsp_transport_free):
19103 * gst-libs/gst/rtsp/gstrtsptransport.h:
19104 * gst-libs/gst/rtsp/gstrtspurl.c: (gst_rtsp_url_parse),
19105 (gst_rtsp_url_free), (gst_rtsp_url_set_port),
19106 (gst_rtsp_url_get_port), (gst_rtsp_url_get_request_uri):
19107 * gst-libs/gst/rtsp/gstrtspurl.h:
19108 * gst-libs/gst/sdp/Makefile.am:
19109 * gst-libs/gst/sdp/gstsdp.h:
19110 * gst-libs/gst/sdp/gstsdpmessage.c: (gst_sdp_origin_init),
19111 (gst_sdp_connection_init), (gst_sdp_bandwidth_init),
19112 (gst_sdp_time_init), (gst_sdp_zone_init), (gst_sdp_key_init),
19113 (gst_sdp_attribute_init), (gst_sdp_message_new),
19114 (gst_sdp_message_init), (gst_sdp_message_uninit),
19115 (gst_sdp_message_free), (gst_sdp_media_new), (gst_sdp_media_init),
19116 (gst_sdp_media_uninit), (gst_sdp_media_free),
19117 (gst_sdp_message_set_origin), (gst_sdp_message_get_origin),
19118 (gst_sdp_message_set_connection), (gst_sdp_message_get_connection),
19119 (gst_sdp_message_add_bandwidth), (gst_sdp_message_add_time),
19120 (gst_sdp_message_add_zone), (gst_sdp_message_set_key),
19121 (gst_sdp_message_get_key), (gst_sdp_message_get_attribute_val_n),
19122 (gst_sdp_message_get_attribute_val),
19123 (gst_sdp_message_add_attribute), (gst_sdp_message_add_media),
19124 (gst_sdp_media_add_attribute), (gst_sdp_media_add_bandwidth),
19125 (gst_sdp_media_add_format), (gst_sdp_media_get_attribute),
19126 (gst_sdp_media_get_attribute_val_n),
19127 (gst_sdp_media_get_attribute_val), (gst_sdp_media_get_format),
19128 (read_string), (read_string_del), (gst_sdp_parse_line),
19129 (gst_sdp_message_parse_buffer), (print_media),
19130 (gst_sdp_message_dump):
19131 * gst-libs/gst/sdp/gstsdpmessage.h:
19132 * pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
19133 Move SDP and RTSP from helper objects in -good to a reusable library.
19134 Use a proper gst_ namespace.
19136 2007-07-23 18:42:22 +0000 Sebastian Dröge <slomo@circular-chaos.org>
19138 ext/vorbis/vorbisdec.c: Use the new buffer clipping function from gstaudio here.
19139 Original commit message from CVS:
19140 * ext/vorbis/vorbisdec.c: (vorbis_dec_push_forward),
19141 (vorbis_dec_flush_decode):
19142 Use the new buffer clipping function from gstaudio here.
19144 2007-07-23 18:26:09 +0000 Sebastian Dröge <slomo@circular-chaos.org>
19146 API: Add buffer clipping function for raw audio buffers. Fixes #456656.
19147 Original commit message from CVS:
19148 * docs/libs/gst-plugins-base-libs-sections.txt:
19149 * gst-libs/gst/audio/audio.c: (gst_audio_buffer_clip):
19150 * gst-libs/gst/audio/audio.h:
19151 * tests/check/libs/audio.c: (GST_START_TEST), (audio_suite):
19152 API: Add buffer clipping function for raw audio buffers. Fixes #456656.
19153 Also add deprecation guards for gst_audio_structure_set_int() to the
19156 2007-07-23 14:45:16 +0000 Stefan Kost <ensonic@users.sourceforge.net>
19158 docs/libs/gst-plugins-base-libs-sections.txt: Cleanup the docs.
19159 Original commit message from CVS:
19160 * docs/libs/gst-plugins-base-libs-sections.txt:
19163 2007-07-23 11:18:35 +0000 Dan Williams <dcbw@redhat.com>
19165 gst/playback/gstplaybasebin.c: Don't return NULL when querying the stream info value array but instead return an empt...
19166 Original commit message from CVS:
19167 Patch by: Dan Williams <dcbw at redhat dot com>
19168 * gst/playback/gstplaybasebin.c:
19169 (gst_play_base_bin_get_streaminfo_value_array):
19170 Don't return NULL when querying the stream info value array but instead
19171 return an empty array. Fixes #459204.
19173 2007-07-23 10:41:18 +0000 Tim-Philipp Müller <tim@centricular.net>
19175 gst/playback/gsturidecodebin.c: Init debug category before using it.
19176 Original commit message from CVS:
19177 * gst/playback/gsturidecodebin.c:
19178 Init debug category before using it.
19180 2007-07-21 09:56:09 +0000 Jan Schmidt <thaytan@mad.scientist.com>
19182 gst-libs/gst/interfaces/mixer.h: Add padding vars in place of the signal pointers when building with DISABLE_DEPRECAT...
19183 Original commit message from CVS:
19184 * gst-libs/gst/interfaces/mixer.h:
19185 Add padding vars in place of the signal pointers
19186 when building with DISABLE_DEPRECATED so that the
19187 interface structure doesn't change size.
19189 2007-07-21 09:21:12 +0000 Marc-Andre Lureau <marcandre.lureau@gmail.com>
19192 Original commit message from CVS:
19193 * docs/libs/gst-plugins-base-libs-sections.txt:
19194 * ext/alsa/gstalsamixer.c:
19195 * ext/alsa/gstalsamixer.h:
19196 * ext/alsa/gstalsamixerelement.c:
19197 * ext/alsa/gstalsamixertrack.c:
19198 * gst-libs/gst/interfaces/mixer.c:
19199 * gst-libs/gst/interfaces/mixer.h:
19200 * gst-libs/gst/interfaces/mixeroptions.c:
19201 * gst-libs/gst/interfaces/mixeroptions.h:
19202 * gst-libs/gst/interfaces/mixertrack.c:
19203 * gst-libs/gst/interfaces/mixertrack.h:
19204 * tests/check/Makefile.am:
19205 * tests/check/libs/mixer.c:
19206 Patch By: Marc-Andre Lureau <marcandre.lureau@gmail.com>
19208 Add support for notifying mixer changes on the message bus, and
19209 implement it in alsamixer.
19210 API: gst_mixer_get_mixer_flags
19211 API: gst_mixer_message_parse_mute_toggled
19212 API: gst_mixer_message_parse_record_toggled
19213 API: gst_mixer_message_parse_volume_changed
19214 API: gst_mixer_message_parse_option_changed
19215 API: GstMixerMessageType
19218 2007-07-20 16:09:03 +0000 Michael Smith <msmith@xiph.org>
19220 sys/xvimage/xvimagesink.c: xcontext->im_format is only for testing XShm support (as the header file comments document...
19221 Original commit message from CVS:
19222 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_new),
19223 (gst_xvimagesink_get_xv_support), (gst_xvimagesink_setcaps):
19224 xcontext->im_format is only for testing XShm support (as the header
19225 file comments document). Use xvimage->im_format for everything else.
19226 Avoids spurious warnings on buffer allocation before setcaps.
19228 2007-07-20 07:22:15 +0000 Stefan Kost <ensonic@users.sourceforge.net>
19230 tests/: We should use $(LIBM).
19231 Original commit message from CVS:
19232 * tests/examples/volume/Makefile.am:
19233 * tests/icles/Makefile.am:
19234 We should use $(LIBM).
19236 2007-07-20 06:13:21 +0000 Stefan Kost <ensonic@users.sourceforge.net>
19238 tests/icles/Makefile.am: This needs -lm.
19239 Original commit message from CVS:
19240 * tests/icles/Makefile.am:
19243 2007-07-18 07:35:32 +0000 Stefan Kost <ensonic@users.sourceforge.net>
19245 Add stdlib include (free, atoi, exit).
19246 Original commit message from CVS:
19247 * examples/app/appsrc_ex.c:
19248 * examples/switch/switcher.c:
19249 * ext/neon/gstneonhttpsrc.c:
19250 * ext/timidity/gstwildmidi.c:
19251 * ext/x264/gstx264enc.c:
19252 * gst/mve/mveaudioenc.c: (mve_compress_audio):
19253 * gst/rtpmanager/gstrtpclient.c:
19254 * gst/rtpmanager/gstrtpjitterbuffer.c:
19255 * gst/spectrum/demo-audiotest.c:
19256 * gst/spectrum/demo-osssrc.c:
19257 * sys/dvb/gstdvbsrc.c:
19258 Add stdlib include (free, atoi, exit).
19260 2007-07-16 10:10:28 +0000 Wim Taymans <wim.taymans@gmail.com>
19262 gst-libs/gst/rtp/gstbasertppayload.c: Don't break ABI, restore previous ranges. Keep the default random selection of ...
19263 Original commit message from CVS:
19264 * gst-libs/gst/rtp/gstbasertppayload.c:
19265 (gst_basertppayload_class_init), (gst_basertppayload_init),
19266 (gst_basertppayload_set_property),
19267 (gst_basertppayload_get_property):
19268 Don't break ABI, restore previous ranges. Keep the default random
19269 selection of timestamp and seqnum offset but as soon as the app sets a
19270 specific value, use that one.
19272 2007-07-14 18:33:15 +0000 Bastien Nocera <hadess@hadess.net>
19274 sys/xvimage/xvimagesink.*: Add option to turn off double-buffering for debugging purposes.
19275 Original commit message from CVS:
19276 Patch by: Bastien Nocera <hadess at hadess dot net>
19277 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support),
19278 (gst_xvimagesink_set_property), (gst_xvimagesink_get_property),
19279 (gst_xvimagesink_init), (gst_xvimagesink_class_init):
19280 * sys/xvimage/xvimagesink.h:
19281 Add option to turn off double-buffering for debugging purposes.
19284 2007-07-14 18:20:41 +0000 Jorn Baayen <jorn@openedhand.com>
19286 sys/: add 'handle-expose' property. Useful for video widgets which may want to be in control of Expose behaviour. Fix...
19287 Original commit message from CVS:
19288 Patch by: Jorn Baayen <jorn at openedhand dot com>
19289 * sys/ximage/ximagesink.c: (gst_ximagesink_handle_xevents),
19290 (gst_ximagesink_set_property), (gst_ximagesink_get_property),
19291 (gst_ximagesink_init), (gst_ximagesink_class_init):
19292 * sys/ximage/ximagesink.h:
19293 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_handle_xevents),
19294 (gst_xvimagesink_set_property), (gst_xvimagesink_get_property),
19295 (gst_xvimagesink_init), (gst_xvimagesink_class_init):
19296 * sys/xvimage/xvimagesink.h:
19297 add 'handle-expose' property. Useful for video widgets which may want to
19298 be in control of Expose behaviour. Fixes #380625
19300 2007-07-14 17:23:42 +0000 Wim Taymans <wim.taymans@gmail.com>
19302 gst-libs/gst/rtp/gstbasertppayload.*: Fix ranges of rtp payloader properties so that the full range can be used in ad...
19303 Original commit message from CVS:
19304 * gst-libs/gst/rtp/gstbasertppayload.c:
19305 (gst_basertppayload_class_init), (gst_basertppayload_init),
19306 (gst_basertppayload_event), (gst_basertppayload_push),
19307 (gst_basertppayload_set_property),
19308 (gst_basertppayload_get_property),
19309 (gst_basertppayload_change_state):
19310 * gst-libs/gst/rtp/gstbasertppayload.h:
19311 Fix ranges of rtp payloader properties so that the full range can be
19312 used in addition to -1 (random).
19313 Fix wrong seqnum reporting in caps.
19316 2007-07-13 18:12:19 +0000 Wim Taymans <wim.taymans@gmail.com>
19318 gst/videorate/gstvideorate.c: Use boilerplate.
19319 Original commit message from CVS:
19320 * gst/videorate/gstvideorate.c: (gst_video_rate_init),
19321 (gst_video_rate_query):
19323 Add latency query, might not be perfect yet but already works a lot
19324 better. Fixes #442557.
19326 2007-07-13 16:05:17 +0000 Jan Schmidt <thaytan@mad.scientist.com>
19328 sys/xvimage/xvimagesink.*: After a caps change, redraw our borders to avoid garbage left there when the image format ...
19329 Original commit message from CVS:
19330 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_put),
19331 (gst_xvimagesink_setcaps):
19332 * sys/xvimage/xvimagesink.h:
19333 After a caps change, redraw our borders to avoid garbage left there
19334 when the image format changes to a smaller size, like 16:9 -> 4:3
19335 Also, hold the flow_lock a bit longer in the set_caps while we're
19336 fiddling with the xcontext.
19338 2007-07-13 16:02:23 +0000 Jan Schmidt <thaytan@mad.scientist.com>
19340 Remove bogus check for libcheck, since we check for gstreamer-check and it pulls in the required info from there, and...
19341 Original commit message from CVS:
19344 * tests/Makefile.am:
19345 Remove bogus check for libcheck, since we check for
19346 gstreamer-check and it pulls in the required info from there, and we
19347 weren't actually _using_ the information for libcheck ourselves
19350 2007-07-13 15:52:02 +0000 Jan Schmidt <thaytan@mad.scientist.com>
19352 gst/ffmpegcolorspace/gstffmpegcodecmap.c: Fix the r_mask test for RGBA32 on little-endian.
19353 Original commit message from CVS:
19354 * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
19355 (gst_ffmpeg_caps_to_pixfmt):
19356 Fix the r_mask test for RGBA32 on little-endian.
19357 Fix a stupid typo that would have obviously broken
19358 compilation on big-endian, if anyone was testing.
19360 2007-07-12 15:02:43 +0000 Wim Taymans <wim.taymans@gmail.com>
19362 gst/videotestsrc/videotestsrc.*: Add alpha to the color struct.
19363 Original commit message from CVS:
19364 * gst/videotestsrc/videotestsrc.c: (paint_hline_AYUV),
19365 (paint_hline_str4):
19366 * gst/videotestsrc/videotestsrc.h:
19367 Add alpha to the color struct.
19368 Use a default alpha value of 255 instead of 128.
19370 2007-07-12 12:01:20 +0000 Wim Taymans <wim.taymans@gmail.com>
19372 gst/playback/gstplaybasebin.c: Clear the dynamic pads counter when starting a new uri. This makes reusing playbin wor...
19373 Original commit message from CVS:
19374 * gst/playback/gstplaybasebin.c: (no_more_pads_full),
19376 Clear the dynamic pads counter when starting a new uri. This makes
19377 reusing playbin work again.
19380 2007-07-12 11:13:32 +0000 Stefan Kost <ensonic@users.sourceforge.net>
19382 configure.ac: Use pkg-config to locate check.
19383 Original commit message from CVS:
19385 Use pkg-config to locate check.
19387 2007-07-11 23:12:12 +0000 Tim-Philipp Müller <tim@centricular.net>
19389 Fix 'make check' build against core CVS.
19390 Original commit message from CVS:
19392 * tests/check/elements/volume.c: (GST_START_TEST):
19393 Fix 'make check' build against core CVS.
19395 2007-07-10 20:46:41 +0000 Stefan Kost <ensonic@users.sourceforge.net>
19397 gst-libs/gst/: Make gtk-doc happy.
19398 Original commit message from CVS:
19399 * gst-libs/gst/interfaces/propertyprobe.c:
19400 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
19401 * gst-libs/gst/tag/gstvorbistag.c:
19402 Make gtk-doc happy.
19404 2007-07-08 13:07:38 +0000 Tim-Philipp Müller <tim@centricular.net>
19406 gst-libs/gst/audio/gstbaseaudiosink.c: Quick hack to make audiosinks stop at EOS when operating in pull-mode; needs t...
19407 Original commit message from CVS:
19408 * gst-libs/gst/audio/gstbaseaudiosink.c:
19409 (gst_base_audio_sink_callback):
19410 Quick hack to make audiosinks stop at EOS when operating in
19411 pull-mode; needs to be fixed properly some day.
19413 2007-07-06 18:19:39 +0000 Stefan Kost <ensonic@users.sourceforge.net>
19415 docs/libs/gst-plugins-base-libs-sections.txt: Fix location of includes in the docs.
19416 Original commit message from CVS:
19417 * docs/libs/gst-plugins-base-libs-sections.txt:
19418 Fix location of includes in the docs.
19420 2007-07-06 11:40:45 +0000 Jan Schmidt <thaytan@mad.scientist.com>
19422 gst/ffmpegcolorspace/: Add 2 new pixel formats - ABGR32 and ARGB32, which are reflections of the existing BGRA32 and ...
19423 Original commit message from CVS:
19424 * gst/ffmpegcolorspace/avcodec.h:
19425 * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
19426 (gst_ffmpeg_pixfmt_to_caps), (gst_ffmpeg_caps_to_pixfmt),
19427 (gst_ffmpegcsp_avpicture_fill):
19428 * gst/ffmpegcolorspace/imgconvert.c: (img_convert),
19429 (img_get_alpha_info):
19430 Add 2 new pixel formats - ABGR32 and ARGB32, which are reflections
19431 of the existing BGRA32 and RGBA32 formats with the alpha at the other
19432 end of the word. Partially fixes #451908
19434 2007-07-05 08:43:30 +0000 Stefan Kost <ensonic@users.sourceforge.net>
19436 docs/: Simplify --extra-dir as gtkdoc scans recursively.
19437 Original commit message from CVS:
19438 * docs/libs/Makefile.am:
19439 * docs/plugins/Makefile.am:
19440 Simplify --extra-dir as gtkdoc scans recursively.
19442 2007-07-03 11:52:47 +0000 Wim Taymans <wim.taymans@gmail.com>
19444 gst/adder/gstadder.c: Make getcaps more robust by not using the proxycaps function. This makes sure that we don't end...
19445 Original commit message from CVS:
19446 * gst/adder/gstadder.c: (gst_adder_sink_getcaps),
19447 (gst_adder_request_new_pad):
19448 Make getcaps more robust by not using the proxycaps function. This makes
19449 sure that we don't end up recursively calling getcaps upstream.
19452 2007-06-29 17:21:18 +0000 Wim Taymans <wim.taymans@gmail.com>
19454 gst/audioconvert/audioconvert.c: Include math.h to fix compilation.
19455 Original commit message from CVS:
19456 * gst/audioconvert/audioconvert.c:
19457 Include math.h to fix compilation.
19459 2007-06-29 14:47:42 +0000 Jan Schmidt <thaytan@mad.scientist.com>
19461 gst/ffmpegcolorspace/gstffmpegcodecmap.c: Add a mapping for YUV format "IYU1", which is a 4:1:1 packed pixel format, ...
19462 Original commit message from CVS:
19463 * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
19464 (gst_ffmpeg_pixfmt_to_caps), (gst_ffmpeg_caps_to_pixfmt):
19465 Add a mapping for YUV format "IYU1", which is a 4:1:1 packed pixel
19466 format, as produced by some dc1394 cameras like the iSight.
19467 See http://www.fourcc.org/yuv.php#IYU1
19469 2007-06-28 20:37:58 +0000 Sebastian Dröge <slomo@circular-chaos.org>
19471 gst/audioconvert/: Implement dithering and noise shaping in audioconvert. By default now
19472 Original commit message from CVS:
19473 * gst/audioconvert/Makefile.am:
19474 * gst/audioconvert/audioconvert.c: (audio_convert_get_func_index),
19475 (check_default), (audio_convert_prepare_context),
19476 (audio_convert_clean_context), (audio_convert_convert):
19477 * gst/audioconvert/audioconvert.h:
19478 * gst/audioconvert/gstaudioconvert.c:
19479 (gst_audio_convert_dithering_get_type),
19480 (gst_audio_convert_ns_get_type), (gst_audio_convert_class_init),
19481 (gst_audio_convert_init), (gst_audio_convert_set_caps),
19482 (gst_audio_convert_set_property), (gst_audio_convert_get_property):
19483 * gst/audioconvert/gstaudioconvert.h:
19484 * gst/audioconvert/gstaudioquantize.c:
19485 (gst_audio_quantize_setup_noise_shaping),
19486 (gst_audio_quantize_free_noise_shaping),
19487 (gst_audio_quantize_setup_dither),
19488 (gst_audio_quantize_free_dither),
19489 (gst_audio_quantize_setup_quantize_func),
19490 (gst_audio_quantize_setup), (gst_audio_quantize_free):
19491 * gst/audioconvert/gstaudioquantize.h:
19492 Implement dithering and noise shaping in audioconvert. By default now
19493 TPDF dithering (and no noise shaping) will be used when converting
19494 from a higher bit depth to 20 bit depth or smaller, otherwise
19495 everything will be as it is now.
19496 For the last audioconvert in a pipeline it would make sense to
19497 use some kind of noise shaping, enabling it by default for all
19498 conversions would give undesired results though. Fixes #360246.
19499 * tests/check/elements/audioconvert.c: (setup_audioconvert),
19501 Adjust unit test for the new audioconvert.
19503 2007-06-28 11:06:56 +0000 Wim Taymans <wim.taymans@gmail.com>
19505 gst/playback/gstqueue2.c: Use other metrics as well when estimating the buffer level.
19506 Original commit message from CVS:
19507 * gst/playback/gstqueue2.c: (apply_segment), (update_buffering):
19508 Use other metrics as well when estimating the buffer level.
19510 2007-06-28 10:21:19 +0000 Wim Taymans <wim.taymans@gmail.com>
19512 gst/playback/gstplaybasebin.c: Small debug improvement.
19513 Original commit message from CVS:
19514 * gst/playback/gstplaybasebin.c: (make_decoder), (setup_source):
19515 Small debug improvement.
19516 * gst/playback/gstqueue2.c: (apply_segment), (update_buffering),
19518 Tweak the rate estimation period.
19519 When calculating the buffer filledness in rate estimation mode, don't
19520 mix it with other metrics.
19522 2007-06-28 09:46:11 +0000 Wim Taymans <wim.taymans@gmail.com>
19524 gst/playback/gstdecodebin2.c: When creating the groups, allow for a 5 second, unlimited buffers preroll phase after w...
19525 Original commit message from CVS:
19526 * gst/playback/gstdecodebin2.c: (gst_decode_group_new),
19527 (gst_decode_group_expose), (gst_decode_group_free), (add_fakesink):
19528 When creating the groups, allow for a 5 second, unlimited buffers
19529 preroll phase after which we expose the group.
19530 When the group is exposed, use a small number of buffers up to a 2
19531 second limit. Also disconnect the overrun signal from multiqueue when we
19532 exposed the group because it is not needed anymore.
19534 2007-06-27 22:30:19 +0000 Tim-Philipp Müller <tim@centricular.net>
19536 gst-libs/gst/tag/tags.c: Don't pass trailing zeroes in fixed-size string arrays in ID3v1 tags to utf8-validate; fixes...
19537 Original commit message from CVS:
19538 * gst-libs/gst/tag/tags.c: (gst_tag_freeform_string_to_utf8):
19539 Don't pass trailing zeroes in fixed-size string arrays in ID3v1 tags
19540 to utf8-validate; fixes recognition of ID3v1 tags in UTF-8 encoding
19541 (#451707); also, output some debugging info when dealing with
19543 * tests/check/libs/tag.c: (GST_START_TEST), (tag_suite):
19544 Add unit test for the above.
19546 2007-06-27 12:55:20 +0000 Tim-Philipp Müller <tim@centricular.net>
19548 gst-libs/gst/pbutils/descriptions.c: Add description for Windows Media RTP caps.
19549 Original commit message from CVS:
19550 * gst-libs/gst/pbutils/descriptions.c: (caps_are_rtp_caps):
19551 Add description for Windows Media RTP caps.
19552 * gst-libs/gst/pbutils/missing-plugins.c: (copy_and_clean_caps):
19553 Remove RTP fields that don't define the format from caps.
19555 2007-06-27 10:14:03 +0000 Tim-Philipp Müller <tim@centricular.net>
19557 ext/vorbis/vorbisdec.c: Skip empty buffers, but not empty header buffers. That way the original vorbisdec unit test s...
19558 Original commit message from CVS:
19559 * ext/vorbis/vorbisdec.c: (vorbis_dec_decode_buffer):
19560 Skip empty buffers, but not empty header buffers. That way the original
19561 vorbisdec unit test still passes (#451145); also, take into account
19562 that those empty packets might carry a granulepos.
19563 * tests/check/Makefile.am:
19564 * tests/check/elements/vorbisdec.c:
19565 (_create_codebook_header_buffer), (_create_audio_buffer),
19566 (GST_START_TEST), (vorbisdec_suite):
19567 Add unit test that sends an empty packet.
19569 2007-06-27 09:49:51 +0000 Wim Taymans <wim.taymans@gmail.com>
19571 ext/vorbis/vorbisdec.c: Don't error out on 0-sized packets, just emit a warning because this is not a fatal error. Fi...
19572 Original commit message from CVS:
19573 * ext/vorbis/vorbisdec.c: (vorbis_dec_decode_buffer):
19574 Don't error out on 0-sized packets, just emit a warning because this is
19575 not a fatal error. Fixes #451145.
19577 2007-06-25 12:43:01 +0000 Stefan Kost <ensonic@users.sourceforge.net>
19579 docs/plugins/: Update docs with caps info.
19580 Original commit message from CVS:
19581 * docs/plugins/gst-plugins-base-plugins.args:
19582 * docs/plugins/gst-plugins-base-plugins.signals:
19583 * docs/plugins/inspect/plugin-adder.xml:
19584 * docs/plugins/inspect/plugin-alsa.xml:
19585 * docs/plugins/inspect/plugin-audioconvert.xml:
19586 * docs/plugins/inspect/plugin-audiorate.xml:
19587 * docs/plugins/inspect/plugin-audioresample.xml:
19588 * docs/plugins/inspect/plugin-audiotestsrc.xml:
19589 * docs/plugins/inspect/plugin-cdparanoia.xml:
19590 * docs/plugins/inspect/plugin-decodebin.xml:
19591 * docs/plugins/inspect/plugin-decodebin2.xml:
19592 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
19593 * docs/plugins/inspect/plugin-gdp.xml:
19594 * docs/plugins/inspect/plugin-gnomevfs.xml:
19595 * docs/plugins/inspect/plugin-libvisual.xml:
19596 * docs/plugins/inspect/plugin-ogg.xml:
19597 * docs/plugins/inspect/plugin-pango.xml:
19598 * docs/plugins/inspect/plugin-playbin.xml:
19599 * docs/plugins/inspect/plugin-subparse.xml:
19600 * docs/plugins/inspect/plugin-tcp.xml:
19601 * docs/plugins/inspect/plugin-theora.xml:
19602 * docs/plugins/inspect/plugin-typefindfunctions.xml:
19603 * docs/plugins/inspect/plugin-video4linux.xml:
19604 * docs/plugins/inspect/plugin-videorate.xml:
19605 * docs/plugins/inspect/plugin-videoscale.xml:
19606 * docs/plugins/inspect/plugin-videotestsrc.xml:
19607 * docs/plugins/inspect/plugin-volume.xml:
19608 * docs/plugins/inspect/plugin-vorbis.xml:
19609 * docs/plugins/inspect/plugin-ximagesink.xml:
19610 * docs/plugins/inspect/plugin-xvimagesink.xml:
19611 Update docs with caps info.
19613 2007-06-25 12:04:15 +0000 Tim-Philipp Müller <tim@centricular.net>
19615 po/POTFILES.in: Add more files with translatable strings (#450875).
19616 Original commit message from CVS:
19618 Add more files with translatable strings (#450875).
19620 2007-06-23 14:44:07 +0000 Edward Hervey <bilboed@bilboed.com>
19622 ext/ogg/gstoggdemux.c: The chain should be freed if we error out here, else it will leak.
19623 Original commit message from CVS:
19624 * ext/ogg/gstoggdemux.c: (gst_ogg_demux_find_chains):
19625 The chain should be freed if we error out here, else it will leak.
19626 * gst/playback/gstdecodebin.c: (disconnect_unlinked_signals),
19627 (cleanup_decodebin):
19628 Don't forget to *properly* remove the signals, else it will leak.
19630 2007-06-22 14:25:27 +0000 Jan Schmidt <thaytan@mad.scientist.com>
19632 MAINTAINERS: Updating all the maintainers files
19633 Original commit message from CVS:
19635 Updating all the maintainers files
19637 2007-06-21 08:34:46 +0000 Stefan Kost <ensonic@users.sourceforge.net>
19639 tests/examples/seek/seek.c: Destroy and recreate parse-launch based pipeline after stop to be able to play again. Reo...
19640 Original commit message from CVS:
19641 * tests/examples/seek/seek.c: (update_scale), (play_cb), (stop_cb),
19643 Destroy and recreate parse-launch based pipeline after stop to be able
19644 to play again. Reorder some code and add more comments.
19646 2007-06-20 11:09:03 +0000 Wim Taymans <wim@fluendo.com>
19648 gst/playback/gstdecodebin2.c: When handling a delayed-caps notification case, mark the group as dynamic so that the n...
19649 Original commit message from CVS:
19650 * gst/playback/gstdecodebin2.c: (analyze_new_pad):
19651 When handling a delayed-caps notification case, mark
19652 the group as dynamic so that the nbdynamic count is
19653 incremented and decremented correctly. Fixes: #449156
19654 Patch by: Wim Taymans <wim@fluendo.com>
19656 2007-06-19 19:13:04 +0000 Andy Wingo <wingo@pobox.com>
19659 * gst-libs/gst/audio/gstbaseaudiosink.c:
19660 * win32/common/config.h:
19661 gst-libs/gst/audio/gstbaseaudiosink.c
19662 Original commit message from CVS:
19663 2007-06-19 Andy Wingo <wingo@pobox.com>
19664 * gst-libs/gst/audio/gstbaseaudiosink.c
19665 (gst_base_audio_sink_init): Enable pull-mode operation.
19667 2007-06-19 09:34:35 +0000 Michael Smith <msmith@xiph.org>
19669 gst-libs/gst/riff/riff-media.c: Change minimum rate back to 1000 to allow low-sample-rate wav files to play back.
19670 Original commit message from CVS:
19671 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
19672 Change minimum rate back to 1000 to allow low-sample-rate wav files
19675 2007-06-17 17:27:09 +0000 Thomas Vander Stichele <thomas@apestaart.org>
19677 po/vi.po: Update translations.
19678 Original commit message from CVS:
19680 Update translations.
19682 2007-06-16 03:42:14 +0000 David Schleef <ds@schleef.org>
19684 gst/playback/gstqueue2.c: Fix compile error from ignored return value.
19685 Original commit message from CVS:
19686 * gst/playback/gstqueue2.c:
19687 Fix compile error from ignored return value.
19689 2007-06-15 15:23:36 +0000 Michael Smith <msmith@xiph.org>
19691 gst/videoscale/vs_4tap.c: Update tmpbuf for all neccesary rows, not just one, as is required when downscaling.
19692 Original commit message from CVS:
19693 * gst/videoscale/vs_4tap.c: (vs_image_scale_4tap_Y):
19694 Update tmpbuf for all neccesary rows, not just one, as is required
19698 2007-06-15 11:15:28 +0000 Michael Smith <msmith@xiph.org>
19700 tests/check/pipelines/oggmux.c: Add a test that ensures we set DELTA_UNIT on all non-header, non-video buffers, if we...
19701 Original commit message from CVS:
19702 * tests/check/pipelines/oggmux.c: (validate_ogg_page), (is_video),
19703 (eos_buffer_probe):
19704 Add a test that ensures we set DELTA_UNIT on all non-header,
19705 non-video buffers, if we have a video stream.
19706 * ext/ogg/gstoggmux.c: (gst_ogg_mux_queue_pads),
19707 (gst_ogg_mux_process_best_pad):
19708 Move setting delta_pad to earlier, where we inspect all pads, so
19709 that leading audio pages don't get DELTA_UNIT unset if they come
19710 before the first DELTA_UNIT from video pages. Fixes the newly-added
19711 test. Fixes #385527.
19713 2007-06-14 19:53:27 +0000 Tim-Philipp Müller <tim@centricular.net>
19715 tests/check/pipelines/streamheader.c: Disable test_multifdsink_gdp_vorbisenc() on ppc64 since it fails on the p5-ppc6...
19716 Original commit message from CVS:
19717 * tests/check/pipelines/streamheader.c: (streamheader_suite):
19718 Disable test_multifdsink_gdp_vorbisenc() on ppc64 since it
19719 fails on the p5-ppc64 build bot and the failure looks like it is due
19720 to the same issue as #348114, ie. a compiler bug.
19722 2007-06-13 18:20:57 +0000 Edward Hervey <bilboed@bilboed.com>
19724 gst/playback/gstqueue2.c: Fix build on MacOSX.
19725 Original commit message from CVS:
19726 * gst/playback/gstqueue2.c: (gst_queue_create_read):
19727 Fix build on MacOSX.
19729 2007-06-13 09:01:32 +0000 Wim Taymans <wim.taymans@gmail.com>
19731 ext/ogg/gstoggdemux.c: Fix compilation on mingw. Fixes #446972.
19732 Original commit message from CVS:
19733 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_packet),
19734 (gst_ogg_demux_perform_seek), (gst_ogg_demux_read_chain):
19735 Fix compilation on mingw. Fixes #446972.
19737 2007-06-12 08:38:06 +0000 Wim Taymans <wim.taymans@gmail.com>
19739 gst/playback/gstqueue2.c: Fix a division by zero when the max percent is <= 0. Fixes #446572. also update the bufferi...
19740 Original commit message from CVS:
19741 Patches by: Thiago Sousa Santos <thiagossantos at gmail dot com>
19742 * gst/playback/gstqueue2.c: (update_buffering),
19743 (gst_queue_locked_enqueue):
19744 Fix a division by zero when the max percent is <= 0. Fixes #446572.
19745 also update the buffering status when receiving events. Fixes #446551.
19747 2007-06-11 11:32:26 +0000 Thiago Sousa Santos <thiagossantos@gmail.com>
19749 gst/playback/gstqueue2.c: Wait for preroll before attempting to forward a duration query upstream.
19750 Original commit message from CVS:
19751 Based on patch by: Thiago Sousa Santos <thiagossantos at gmail dot com>
19752 * gst/playback/gstqueue2.c: (gst_queue_peer_query),
19753 (gst_queue_handle_src_query):
19754 Wait for preroll before attempting to forward a duration query upstream.
19757 2007-06-07 21:08:38 +0000 Sébastien Moutte <sebastien@moutte.net>
19759 gst-libs/gst/rtp/gstbasertpdepayload.c: Use G_GINT64_CONSTANT macro for int64 constant.
19760 Original commit message from CVS:
19761 * gst-libs/gst/rtp/gstbasertpdepayload.c:
19762 (gst_base_rtp_depayload_set_gst_timestamp):
19763 Use G_GINT64_CONSTANT macro for int64 constant.
19764 * win32/common/libgstinterfaces.def:
19765 * win32/common/libgsttag.def:
19766 Add new exported functions.
19768 2007-06-07 14:25:32 +0000 Tim-Philipp Müller <tim@centricular.net>
19770 ext/ogg/gstoggmux.c: The BOS page of the first Dirac video stream needs to come before the BOS page of any Vorbis str...
19771 Original commit message from CVS:
19772 * ext/ogg/gstoggmux.c: (gst_ogg_mux_send_headers):
19773 The BOS page of the first Dirac video stream needs to come before
19774 the BOS page of any Vorbis streams or other audio streams, just like
19777 2007-06-07 09:11:27 +0000 Wim Taymans <wim.taymans@gmail.com>
19779 gst/playback/gstqueue2.c: Fix compilation.
19780 Original commit message from CVS:
19781 * gst/playback/gstqueue2.c: (gst_queue_get_range):
19784 2007-06-06 13:36:26 +0000 Thiago Sousa Santos <thiagossantos@gmail.com>
19786 gst/playback/gstqueue2.c: Add pull based scheduling and fix some deadlocks. Fixes #444523.
19787 Original commit message from CVS:
19788 Patch by: Thiago Sousa Santos <thiagossantos at gmail dot com>
19789 * gst/playback/gstqueue2.c: (gst_queue_init),
19790 (gst_queue_handle_sink_event), (gst_queue_chain),
19791 (gst_queue_get_range), (gst_queue_src_checkgetrange_function),
19792 (gst_queue_sink_activate_push), (gst_queue_src_activate_push),
19793 (gst_queue_src_activate_pull):
19794 Add pull based scheduling and fix some deadlocks. Fixes #444523.
19795 Does not yet completely work because duration queries upstream won't
19798 2007-06-06 09:08:50 +0000 Wim Taymans <wim.taymans@gmail.com>
19800 Some more fseeko checks.
19801 Original commit message from CVS:
19803 * gst/playback/gstqueue2.c: (gst_queue_create_read):
19804 Some more fseeko checks.
19806 2007-06-06 08:01:42 +0000 Wim Taymans <wim.taymans@gmail.com>
19808 configure.ac: check for large file support.
19809 Original commit message from CVS:
19811 check for large file support.
19813 2007-06-05 21:36:11 +0000 Sven Arvidsson <sa@whiz.se>
19815 gst/subparse/gstsubparse.*: Add support for SubViewer version 1 and 2 subtitles. Fixes #394061.
19816 Original commit message from CVS:
19817 Based on a patch by Sven Arvidsson <sa at whiz dot se>:
19818 * gst/subparse/gstsubparse.c: (parse_subrip),
19819 (subviewer_unescape_newlines), (parse_subviewer),
19820 (gst_sub_parse_data_format_autodetect),
19821 (gst_sub_parse_format_autodetect), (gst_subparse_type_find):
19822 * gst/subparse/gstsubparse.h:
19823 Add support for SubViewer version 1 and 2 subtitles. Fixes #394061.
19824 * tests/check/elements/subparse.c: (GST_START_TEST),
19826 Add a unit test for both SubViewer formats.
19828 2007-06-05 17:08:04 +0000 Michael Smith <msmith@xiph.org>
19830 gst/audiotestsrc/gstaudiotestsrc.c: Don't overflow intermediate values when seeking to large time values in audiotest...
19831 Original commit message from CVS:
19832 * gst/audiotestsrc/gstaudiotestsrc.c: (gst_audio_test_src_do_seek):
19833 Don't overflow intermediate values when seeking to large time values
19836 2007-06-05 17:02:13 +0000 Wim Taymans <wim.taymans@gmail.com>
19838 gst/playback/gstqueue2.c: Include stdio to define fseeko.
19839 Original commit message from CVS:
19840 * gst/playback/gstqueue2.c: (gst_queue_have_data),
19841 (gst_queue_create_read), (gst_queue_read_item_from_file),
19842 (gst_queue_open_temp_location_file), (gst_queue_locked_enqueue):
19843 Include stdio to define fseeko.
19845 2007-06-05 16:37:09 +0000 Edward Hervey <edward@fluendo.com>
19847 sys/v4l/gstv4lsrc.c: Make v4lsrc output segments in GST_FORMAT_TIME. Fixes #442553.
19848 Original commit message from CVS:
19849 Patch by: Edward Hervey <edward@fluendo.com>
19850 * sys/v4l/gstv4lsrc.c: (gst_v4lsrc_init), (gst_v4lsrc_fixate),
19851 (gst_v4lsrc_query):
19852 Make v4lsrc output segments in GST_FORMAT_TIME. Fixes #442553.
19854 2007-06-05 16:20:44 +0000 Tim-Philipp Müller <tim@centricular.net>
19856 gst-libs/gst/riff/: Use gst_tag_utf8_from_freeform_string() from libgsttag instead of our own implementation.
19857 Original commit message from CVS:
19858 * gst-libs/gst/riff/Makefile.am:
19859 * gst-libs/gst/riff/riff-read.c: (gst_riff_parse_info):
19860 Use gst_tag_utf8_from_freeform_string() from libgsttag instead of
19861 our own implementation.
19863 2007-06-05 16:19:30 +0000 Wim Taymans <wim.taymans@gmail.com>
19865 gst-libs/gst/rtp/gstbasertpdepayload.c: Handle timestamp wraparound.
19866 Original commit message from CVS:
19867 * gst-libs/gst/rtp/gstbasertpdepayload.c:
19868 (gst_base_rtp_depayload_setcaps),
19869 (gst_base_rtp_depayload_set_gst_timestamp),
19870 (gst_base_rtp_depayload_change_state):
19871 Handle timestamp wraparound.
19873 2007-06-05 16:17:30 +0000 Wim Taymans <wim.taymans@gmail.com>
19875 gst/playback/gsturidecodebin.c: Make sure we name srcpads uniquely even when using different internal decodebins.
19876 Original commit message from CVS:
19877 * gst/playback/gsturidecodebin.c: (no_more_pads_full),
19878 (new_decoded_pad), (remove_pads), (make_decoder), (setup_source),
19879 (gst_uri_decode_bin_change_state):
19880 Make sure we name srcpads uniquely even when using different internal
19882 Signal no-more-pads when no more dynamic elements exist.
19883 Remove pads on cleanup.
19885 2007-06-05 16:14:23 +0000 Thiago Sousa Santos <thiagossantos@gmail.com>
19887 gst/playback/gstqueue2.c: Add support for filebased buffering. Fixes #441264.
19888 Original commit message from CVS:
19889 Based on patch by: Thiago Sousa Santos <thiagossantos at gmail dot com>
19890 * gst/playback/gstqueue2.c: (gst_queue_class_init),
19891 (gst_queue_init), (gst_queue_finalize),
19892 (gst_queue_write_buffer_to_file), (gst_queue_have_data),
19893 (gst_queue_create_read), (gst_queue_read_item_from_file),
19894 (gst_queue_open_temp_location_file),
19895 (gst_queue_close_temp_location_file), (gst_queue_locked_flush),
19896 (gst_queue_locked_enqueue), (gst_queue_locked_dequeue),
19897 (gst_queue_is_empty), (gst_queue_is_filled),
19898 (gst_queue_change_state), (gst_queue_set_temp_location),
19899 (gst_queue_set_property):
19900 Add support for filebased buffering. Fixes #441264.
19902 2007-06-05 16:05:19 +0000 Wim Taymans <wim.taymans@gmail.com>
19904 gst/playback/gstdecodebin2.c: Add support for delayed caps fixation when autoplugging.
19905 Original commit message from CVS:
19906 * gst/playback/gstdecodebin2.c: (gst_decode_bin_factory_filter),
19907 (analyze_new_pad), (connect_pad), (expose_pad), (caps_notify_cb),
19908 (caps_notify_group_cb), (gst_decode_group_new),
19909 (gst_decode_group_free):
19910 Add support for delayed caps fixation when autoplugging.
19911 Optimize cases where a multiqueue is not needed/wanted, like right after
19912 anything that is not a demuxer.
19914 2007-06-05 16:02:57 +0000 Wim Taymans <wim.taymans@gmail.com>
19916 ext/ogg/gstoggdemux.c: consideratly speedup ogg chain detection by not trying to find a base timestamp for skeleton s...
19917 Original commit message from CVS:
19918 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_parse_skeleton_fisbone),
19919 (gst_ogg_pad_submit_packet), (gst_ogg_demux_perform_seek),
19920 (gst_ogg_demux_read_chain), (gst_ogg_demux_collect_chain_info):
19921 consideratly speedup ogg chain detection by not trying to find a base
19922 timestamp for skeleton streams.
19924 2007-06-05 16:00:33 +0000 Wim Taymans <wim.taymans@gmail.com>
19926 gst/tcp/gstmultifdsink.*: Add support for remuve_flush.
19927 Original commit message from CVS:
19928 * gst/tcp/gstmultifdsink.c: (gst_client_status_get_type),
19929 (gst_multi_fd_sink_class_init), (gst_multi_fd_sink_add_full),
19930 (gst_multi_fd_sink_remove_flush),
19931 (gst_multi_fd_sink_remove_client_link),
19932 (gst_multi_fd_sink_handle_client_write),
19933 (gst_multi_fd_sink_handle_clients):
19934 * gst/tcp/gstmultifdsink.h:
19935 Add support for remuve_flush.
19937 2007-06-05 15:59:00 +0000 Wim Taymans <wim.taymans@gmail.com>
19939 Add draft design for forcing keyframes in encoders and implement in theoraenc.
19940 Original commit message from CVS:
19941 * docs/design/draft-keyframe-force.txt:
19942 * ext/theora/theoraenc.c: (theora_enc_sink_event),
19943 (theora_enc_chain):
19944 Add draft design for forcing keyframes in encoders and implement in
19947 2007-06-05 13:22:18 +0000 Jan Schmidt <thaytan@mad.scientist.com>
19949 configure.ac: Back to CVS
19950 Original commit message from CVS:
19954 === release 0.10.13 ===
19956 2007-06-05 12:50:24 +0000 Jan Schmidt <thaytan@mad.scientist.com>
19962 * docs/plugins/gst-plugins-base-plugins.args:
19963 * docs/plugins/inspect/plugin-adder.xml:
19964 * docs/plugins/inspect/plugin-alsa.xml:
19965 * docs/plugins/inspect/plugin-audioconvert.xml:
19966 * docs/plugins/inspect/plugin-audiorate.xml:
19967 * docs/plugins/inspect/plugin-audioresample.xml:
19968 * docs/plugins/inspect/plugin-audiotestsrc.xml:
19969 * docs/plugins/inspect/plugin-cdparanoia.xml:
19970 * docs/plugins/inspect/plugin-decodebin.xml:
19971 * docs/plugins/inspect/plugin-decodebin2.xml:
19972 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
19973 * docs/plugins/inspect/plugin-gdp.xml:
19974 * docs/plugins/inspect/plugin-gnomevfs.xml:
19975 * docs/plugins/inspect/plugin-libvisual.xml:
19976 * docs/plugins/inspect/plugin-ogg.xml:
19977 * docs/plugins/inspect/plugin-pango.xml:
19978 * docs/plugins/inspect/plugin-playbin.xml:
19979 * docs/plugins/inspect/plugin-subparse.xml:
19980 * docs/plugins/inspect/plugin-tcp.xml:
19981 * docs/plugins/inspect/plugin-theora.xml:
19982 * docs/plugins/inspect/plugin-typefindfunctions.xml:
19983 * docs/plugins/inspect/plugin-video4linux.xml:
19984 * docs/plugins/inspect/plugin-videorate.xml:
19985 * docs/plugins/inspect/plugin-videoscale.xml:
19986 * docs/plugins/inspect/plugin-videotestsrc.xml:
19987 * docs/plugins/inspect/plugin-volume.xml:
19988 * docs/plugins/inspect/plugin-vorbis.xml:
19989 * docs/plugins/inspect/plugin-ximagesink.xml:
19990 * docs/plugins/inspect/plugin-xvimagesink.xml:
19991 * gst-plugins-base.doap:
19992 * win32/common/config.h:
19993 * win32/vs6/grammar.dsp:
19994 * win32/vs6/gst_plugins_base.dsw:
19995 * win32/vs6/libgstadder.dsp:
19996 * win32/vs6/libgstaudio.dsp:
19997 * win32/vs6/libgstaudioconvert.dsp:
19998 * win32/vs6/libgstaudiorate.dsp:
19999 * win32/vs6/libgstaudioresample.dsp:
20000 * win32/vs6/libgstaudioscale.dsp:
20001 * win32/vs6/libgstaudiotestsrc.dsp:
20002 * win32/vs6/libgstcdda.dsp:
20003 * win32/vs6/libgstdecodebin.dsp:
20004 * win32/vs6/libgstdecodebin2.dsp:
20005 * win32/vs6/libgstdirectsound.dsp:
20006 * win32/vs6/libgstffmpegcolorspace.dsp:
20007 * win32/vs6/libgstgdp.dsp:
20008 * win32/vs6/libgstinterfaces.dsp:
20009 * win32/vs6/libgstnetbuffer.dsp:
20010 * win32/vs6/libgstogg.dsp:
20011 * win32/vs6/libgstpbutils.dsp:
20012 * win32/vs6/libgstplaybin.dsp:
20013 * win32/vs6/libgstriff.dsp:
20014 * win32/vs6/libgstrtp.dsp:
20015 * win32/vs6/libgstsinesrc.dsp:
20016 * win32/vs6/libgstsubparse.dsp:
20017 * win32/vs6/libgsttag.dsp:
20018 * win32/vs6/libgsttheora.dsp:
20019 * win32/vs6/libgsttypefindfunctions.dsp:
20020 * win32/vs6/libgstutils.dsp:
20021 * win32/vs6/libgstvideo.dsp:
20022 * win32/vs6/libgstvideorate.dsp:
20023 * win32/vs6/libgstvideoscale.dsp:
20024 * win32/vs6/libgstvideotestsrc.dsp:
20025 * win32/vs6/libgstvolume.dsp:
20026 * win32/vs6/libgstvorbis.dsp:
20027 Release 0.10.13 "What's going on?"
20028 Original commit message from CVS:
20029 Release 0.10.13 "What's going on?"
20031 2007-06-05 12:32:03 +0000 Jan Schmidt <thaytan@mad.scientist.com>
20049 Original commit message from CVS:
20052 2007-05-31 17:08:58 +0000 Wim Taymans <wim@fluendo.com>
20054 gst-libs/gst/riff/riff-media.c: In riff, the depth is stored in the size field but it just means that the least signi...
20055 Original commit message from CVS:
20056 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
20057 In riff, the depth is stored in the size field but it just means that
20058 the least significant bits are cleared. We can therefore just play
20059 the sample as if it had a depth == width. Fixes: #440997
20060 Patch by: Wim Taymans <wim@fluendo.com>
20061 Patch by: Sebastian Dröge <slomo@circular-chaos.org>
20063 2007-05-31 16:36:22 +0000 Jan Schmidt <thaytan@mad.scientist.com>
20065 gst-libs/gst/floatcast/floatcast.h: Define inline when needed on win32 builds. Fixes: #441295
20066 Original commit message from CVS:
20067 * gst-libs/gst/floatcast/floatcast.h:
20068 Define inline when needed on win32 builds. Fixes: #441295
20070 2007-05-29 13:38:35 +0000 Wim Taymans <wim.taymans@gmail.com>
20072 gst/playback/gstplaybasebin.c: Stop buffering when the group is commited because the queues filled up.
20073 Original commit message from CVS:
20074 * gst/playback/gstplaybasebin.c: (queue_overrun),
20075 (no_more_pads_full):
20076 Stop buffering when the group is commited because the queues filled up.
20079 2007-05-25 10:07:26 +0000 Jan Schmidt <thaytan@mad.scientist.com>
20081 Revert commits towards #152864 made so far. We'll pick it up again after the 0.10.13 release.
20082 Original commit message from CVS:
20083 * ext/alsa/gstalsamixer.c: (gst_alsa_mixer_ensure_track_list),
20084 (gst_alsa_mixer_free), (gst_alsa_mixer_update),
20085 (gst_alsa_mixer_get_volume), (gst_alsa_mixer_set_volume),
20086 (gst_alsa_mixer_set_mute), (gst_alsa_mixer_set_record),
20087 (gst_alsa_mixer_set_option), (gst_alsa_mixer_get_option):
20088 * ext/alsa/gstalsamixer.h:
20089 * ext/alsa/gstalsamixerelement.c:
20090 (gst_alsa_mixer_element_interface_supported),
20091 (gst_alsa_mixer_element_finalize), (gst_alsa_mixer_element_init),
20092 (gst_alsa_mixer_element_set_property),
20093 (gst_alsa_mixer_element_get_property),
20094 (gst_alsa_mixer_element_change_state):
20095 * ext/alsa/gstalsamixertrack.c: (gst_alsa_mixer_track_update):
20096 * gst-libs/gst/interfaces/mixer.c: (gst_mixer_volume_changed),
20097 (gst_mixer_option_changed):
20098 * gst-libs/gst/interfaces/mixer.h:
20099 Revert commits towards #152864 made so far. We'll pick it up again
20100 after the 0.10.13 release.
20102 2007-05-24 16:22:23 +0000 Wim Taymans <wim.taymans@gmail.com>
20104 gst-libs/gst/audio/gstbaseaudiosink.c: After an interrupt (PAUSED/flush) assume that the next sample should not be al...
20105 Original commit message from CVS:
20106 * gst-libs/gst/audio/gstbaseaudiosink.c:
20107 (gst_base_audio_sink_render):
20108 After an interrupt (PAUSED/flush) assume that the next sample should not
20109 be aligned to the previous sample. Fixes #417992.
20111 2007-05-24 15:16:59 +0000 Tim-Philipp Müller <tim@centricular.net>
20113 gst-libs/gst/riff/riff-media.c: Don't add channels and rate fields to the template caps for audio/x-dts, as wavparse ...
20114 Original commit message from CVS:
20115 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
20116 Don't add channels and rate fields to the template caps for
20117 audio/x-dts, as wavparse might not always be able to set them,
20118 which would then lead to 'caps are not a real subset of the
20119 template caps' warnings.
20121 2007-05-24 11:15:32 +0000 Jan Schmidt <thaytan@mad.scientist.com>
20123 gst/playback/gstplaybasebin.c: Handle unknown or invalid pads without crashing, as might occur if a media file like a...
20124 Original commit message from CVS:
20125 * gst/playback/gstplaybasebin.c: (new_decoded_pad_full):
20126 Handle unknown or invalid pads without crashing, as might occur if
20127 a media file like an mp3 is specified as a subtitle file.
20130 2007-05-24 10:19:54 +0000 Jan Schmidt <thaytan@mad.scientist.com>
20132 gst/playback/gstplaybin.c: Block the subtitle bin output queue before ghosting it and linking, then unblock after. Th...
20133 Original commit message from CVS:
20134 * gst/playback/gstplaybin.c: (add_sink), (dummy_blocked_cb),
20136 Block the subtitle bin output queue before ghosting it and linking,
20137 then unblock after. This avoids spurious not-linked errors caused
20138 by the queue starting up (because it gets linked when it is ghosted).
20141 2007-05-23 15:54:28 +0000 Jan Schmidt <thaytan@mad.scientist.com>
20143 tests/check/elements/playbin.c: Use /dev/zero instead of /dev/urandom to produce an invalid subtitle file. Avoids flu...
20144 Original commit message from CVS:
20145 * tests/check/elements/playbin.c: (test_suburi_error_unknowntype):
20146 Use /dev/zero instead of /dev/urandom to produce an invalid subtitle
20147 file. Avoids flukes where the input gets typefound to some valid but
20150 2007-05-22 15:45:19 +0000 Tim-Philipp Müller <tim@centricular.net>
20152 tests/check/: Add unit test for gnomevfssink seeking and position reporting for file:// URIs.
20153 Original commit message from CVS:
20154 * tests/check/Makefile.am:
20155 * tests/check/elements/.cvsignore:
20156 * tests/check/elements/gnomevfssink.c: (setup_gnomevfssink),
20157 (cleanup_gnomevfssink), (GST_START_TEST), (gnomevfssink_suite):
20158 Add unit test for gnomevfssink seeking and position reporting for
20161 2007-05-22 15:30:26 +0000 Mark Nauwelaerts <manauw@skynet.be>
20163 ext/gnomevfs/gstgnomevfssink.*: see #412648.
20164 Original commit message from CVS:
20165 Patch by: Mark Nauwelaerts <manauw at skynet be>
20166 * ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_init),
20167 (gst_gnome_vfs_sink_open_file), (gst_gnome_vfs_sink_handle_event),
20168 (gst_gnome_vfs_sink_query), (gst_gnome_vfs_sink_render):
20169 * ext/gnomevfs/gstgnomevfssink.h:
20170 Fix position reporting, especially after a seek (from upstream),
20173 2007-05-22 15:04:41 +0000 Tim-Philipp Müller <tim@centricular.net>
20175 ext/cdparanoia/gstcdparanoiasrc.c: Repair umlaut.
20176 Original commit message from CVS:
20177 * ext/cdparanoia/gstcdparanoiasrc.c:
20180 2007-05-22 11:40:31 +0000 Jan Schmidt <thaytan@mad.scientist.com>
20182 gst-libs/gst/riff/riff-media.c: Specify the full valid range for MP3 samplerates. Fixes a regression caused by extra ...
20183 Original commit message from CVS:
20184 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
20185 Specify the full valid range for MP3 samplerates. Fixes a regression
20186 caused by extra header checks since the last release.
20188 2007-05-21 15:32:42 +0000 Mike Smith <msmith@xiph.org>
20190 sys/: Fix a locking-order bug I introduced with my changes the other day.
20191 Original commit message from CVS:
20192 * sys/ximage/ximagesink.c: (gst_ximagesink_handle_xevents):
20193 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_handle_xevents):
20194 Fix a locking-order bug I introduced with my changes the other day.
20195 Patch by Mike Smith.
20197 2007-05-21 15:24:21 +0000 Michael Smith <msmith@xiph.org>
20199 ext/theora/theoradec.c: Don't look inside 0-length packets (which indicate duplicated frames)
20200 Original commit message from CVS:
20201 * ext/theora/theoradec.c: (theora_handle_data_packet):
20202 Don't look inside 0-length packets (which indicate duplicated
20205 2007-05-21 10:25:44 +0000 Wim Taymans <wim.taymans@gmail.com>
20208 Original commit message from CVS:
20209 * ext/cdparanoia/gstcdparanoiasrc.c:
20210 (gst_cd_paranoia_src_read_sector):
20211 * gst-libs/gst/audio/gstbaseaudiosrc.c:
20212 (gst_base_audio_src_create):
20214 * ext/theora/theoradec.c: (theora_dec_sink_event):
20216 * gst-libs/gst/rtp/gstbasertpdepayload.c:
20217 (gst_base_rtp_depayload_set_gst_timestamp):
20219 * gst/playback/gstdecodebin.c: (queue_underrun_cb):
20220 And some debug info when a FIXME path is hit.
20222 2007-05-21 09:45:28 +0000 Wim Taymans <wim.taymans@gmail.com>
20224 gst-libs/gst/rtp/gstbasertpaudiopayload.c: Some cleanups, remove minptime property as it is now in the parent class.
20225 Original commit message from CVS:
20226 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
20227 (gst_base_rtp_audio_payload_class_init),
20228 (gst_base_rtp_audio_payload_init),
20229 (gst_base_rtp_audio_payload_finalize),
20230 (gst_base_rtp_audio_payload_handle_frame_based_buffer),
20231 (gst_base_rtp_audio_payload_handle_sample_based_buffer),
20232 (gst_base_rtp_payload_audio_handle_event):
20233 Some cleanups, remove minptime property as it is now in the parent
20235 Override parent class event function.
20236 * gst-libs/gst/rtp/gstbasertppayload.c:
20237 (gst_basertppayload_class_init), (gst_basertppayload_init),
20238 (gst_basertppayload_event), (gst_basertppayload_set_property),
20239 (gst_basertppayload_get_property):
20240 * gst-libs/gst/rtp/gstbasertppayload.h:
20241 Add min-ptime property.
20242 Add handle-event vmethod. Fixes #415001.
20244 2007-05-18 17:10:03 +0000 Christian Schaller <uraeus@gnome.org>
20246 * gst-plugins-base.spec.in:
20248 Original commit message from CVS:
20251 2007-05-18 15:23:43 +0000 Stefan Kost <ensonic@users.sourceforge.net>
20253 gst-libs/gst/audio/gstbaseaudiosink.c
20254 Original commit message from CVS:
20255 * gst-libs/gst/audio/gstbaseaudiosink.c
20256 (gst_base_audio_sink_change_state):
20257 Fix typo in comment.
20258 * gst/playback/gstdecodebin.c (gst_decode_bin_class_init,
20259 free_dynamics, pad_probe, close_pad_link, try_to_link_1,
20260 get_our_ghost_pad, remove_element_chain, queue_underrun_cb,
20262 * gst/playback/gstplaybin.c (gst_play_bin_set_property,
20263 gen_audio_element, remove_sinks, gst_play_bin_send_event_to_sink):
20264 Remove trailing whitespaces in comments.
20265 * gst/volume/Makefile.am:
20268 2007-05-18 15:10:08 +0000 Marc-Andre Lureau <marcandre.lureau@gmail.com>
20271 * gst-libs/gst/interfaces/mixer.h:
20272 gst-libs/gst/interfaces/mixer.h (mixer_type, option_changed, set_option, get_option, _gst_reserved):
20273 Original commit message from CVS:
20274 patch by: Marc-Andre Lureau <marcandre.lureau@gmail.com>
20275 * gst-libs/gst/interfaces/mixer.h (mixer_type, option_changed,
20276 set_option, get_option, _gst_reserved):
20277 Revert reordering functions (keep ABI).
20279 2007-05-17 17:35:46 +0000 Jan Schmidt <thaytan@mad.scientist.com>
20281 sys/: When we create our own window, indicate that we handle the
20282 Original commit message from CVS:
20283 * sys/ximage/ximagesink.c: (gst_ximagesink_ximage_put),
20284 (gst_ximagesink_xwindow_new), (gst_ximagesink_handle_xevents),
20285 (gst_ximagesink_show_frame):
20286 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_put),
20287 (gst_xvimagesink_xwindow_new), (gst_xvimagesink_handle_xevents),
20288 (gst_xvimagesink_show_frame):
20289 When we create our own window, indicate that we handle the
20290 WM_DELETE client message from the window manager, so that it won't
20291 kill our window (and our app) along with it. Handle ClientMessage,
20292 post an error on the bus, and close the window. Further buffers
20293 arriving will result in a FlowError because the window has been
20296 Clean up the X event handling loop and make them the same for
20297 both xvimagesink and ximagesink while I'm at it.
20299 2007-05-17 16:27:32 +0000 Wim Taymans <wim.taymans@gmail.com>
20301 gst/playback/gstdecodebin2.c: Make decodebin2 autoplug depayloaders too.
20302 Original commit message from CVS:
20303 * gst/playback/gstdecodebin2.c: (gst_decode_bin_factory_filter):
20304 Make decodebin2 autoplug depayloaders too.
20305 * gst/playback/gsturidecodebin.c: (source_new_pad):
20306 Set the newly created decoder in a usable state when autoplugging a
20307 dynamic source such as RTSP.
20309 2007-05-17 16:11:03 +0000 Tim-Philipp Müller <tim@centricular.net>
20311 gst/playback/gststreaminfo.c: Ignore video-codec tag for audio streams and ignore audio-codec tags for video streams....
20312 Original commit message from CVS:
20313 * gst/playback/gststreaminfo.c: (cb_probe):
20314 Ignore video-codec tag for audio streams and ignore audio-codec tags
20315 for video streams. Should make codec name collection a bit more
20316 robust against sloppy demuxers that send tag events containing both
20317 tags down each pad.
20319 2007-05-17 15:22:44 +0000 Wim Taymans <wim.taymans@gmail.com>
20321 gst/playback/gstqueue2.c: Tweak the buffering thresholds a little.
20322 Original commit message from CVS:
20323 * gst/playback/gstqueue2.c: (update_rates):
20324 Tweak the buffering thresholds a little.
20325 Update the buffer size with the previously calculate rate instead of
20326 only when we calculate a new rate so that we get smoother buffering
20328 * gst/playback/Makefile.am:
20329 * gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_base_init),
20330 (gst_uri_decode_bin_class_init), (gst_uri_decode_bin_init),
20331 (gst_uri_decode_bin_finalize), (gst_uri_decode_bin_set_property),
20332 (gst_uri_decode_bin_get_property), (unknown_type),
20333 (add_element_stream), (no_more_pads_full), (no_more_pads),
20334 (source_no_more_pads), (new_decoded_pad), (array_has_value),
20335 (gen_source_element), (has_all_raw_caps), (analyse_source),
20336 (remove_decoders), (make_decoder), (remove_source),
20337 (source_new_pad), (setup_source), (decoder_query_init),
20338 (decoder_query_duration_fold), (decoder_query_duration_done),
20339 (decoder_query_position_fold), (decoder_query_position_done),
20340 (decoder_query_latency_fold), (decoder_query_latency_done),
20341 (decoder_query_seeking_fold), (decoder_query_seeking_done),
20342 (decoder_query_generic_fold), (gst_uri_decode_bin_query),
20343 (gst_uri_decode_bin_change_state), (plugin_init):
20344 New element that intergrates a source, optional buffering element and
20347 2007-05-17 14:17:17 +0000 Tim-Philipp Müller <tim@centricular.net>
20349 configure.ac: Bump libtheora requirement to 1.0alpha5 for the pixformat check (also has a .pc file, so we don't need ...
20350 Original commit message from CVS:
20352 Bump libtheora requirement to 1.0alpha5 for the pixformat check
20353 (also has a .pc file, so we don't need the fallback check any
20354 longer). Fixes #438840.
20356 2007-05-17 13:36:11 +0000 Wim Taymans <wim.taymans@gmail.com>
20358 gst/playback/gstqueue2.c: fix build.
20359 Original commit message from CVS:
20360 * gst/playback/gstqueue2.c: (gst_queue_get_type),
20361 (gst_queue_class_init), (gst_queue_finalize), (update_time_level),
20362 (apply_segment), (apply_buffer), (update_buffering),
20363 (reset_rate_timer), (update_rates), (gst_queue_locked_flush),
20364 (gst_queue_locked_enqueue), (gst_queue_locked_dequeue),
20365 (gst_queue_handle_sink_event), (gst_queue_is_filled),
20366 (gst_queue_chain), (gst_queue_push_one), (gst_queue_loop),
20370 2007-05-17 11:57:44 +0000 Wim Taymans <wim.taymans@gmail.com>
20372 gst/playback/: On our way to playbin2 this is the new network queue that does buffering all by itself using high and ...
20373 Original commit message from CVS:
20374 * gst/playback/Makefile.am:
20375 * gst/playback/gstqueue2.c: (gst_queue_get_type),
20376 (gst_queue_class_init), (gst_queue_init), (gst_queue_finalize),
20377 (gst_queue_getcaps), (gst_queue_bufferalloc),
20378 (gst_queue_acceptcaps), (update_time_level), (apply_segment),
20379 (apply_buffer), (update_buffering), (reset_rate_timer),
20380 (update_rates), (gst_queue_locked_flush),
20381 (gst_queue_locked_enqueue), (gst_queue_locked_dequeue),
20382 (gst_queue_handle_sink_event), (gst_queue_is_empty),
20383 (gst_queue_is_filled), (gst_queue_chain), (gst_queue_push_one),
20384 (gst_queue_loop), (gst_queue_handle_src_event),
20385 (gst_queue_handle_src_query), (gst_queue_sink_activate_push),
20386 (gst_queue_src_activate_push), (gst_queue_change_state),
20387 (gst_queue_set_property), (gst_queue_get_property), (plugin_init):
20388 On our way to playbin2 this is the new network queue that does buffering
20389 all by itself using high and low watermarks. It can also measure up and
20390 downstream bandwidth to optimally size the queue.
20392 2007-05-17 11:16:14 +0000 Michael Smith <msmith@xiph.org>
20394 gst/: Use the segment->last_stop value to calculate the next timestamp to generate after a seek; not the segment->sta...
20395 Original commit message from CVS:
20396 * gst/audiotestsrc/gstaudiotestsrc.c: (gst_audio_test_src_do_seek):
20397 * gst/videotestsrc/gstvideotestsrc.c: (gst_video_test_src_do_seek):
20398 Use the segment->last_stop value to calculate the next timestamp to
20399 generate after a seek; not the segment->start value.
20401 2007-05-15 20:14:06 +0000 David Schleef <ds@schleef.org>
20403 docs/Makefile.am: Install docs even when --disable-gtk-doc is disabled. This matches the behavior of gtk+. Fixes #3...
20404 Original commit message from CVS:
20405 * docs/Makefile.am: Install docs even when --disable-gtk-doc
20406 is disabled. This matches the behavior of gtk+. Fixes #349099.
20408 2007-05-15 17:11:09 +0000 Wim Taymans <wim.taymans@gmail.com>
20410 ext/ogg/gstoggdemux.c: Some more chained streaming ogg timestamp fixes.
20411 Original commit message from CVS:
20412 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_packet),
20413 (gst_ogg_demux_perform_seek), (gst_ogg_demux_handle_page):
20414 Some more chained streaming ogg timestamp fixes.
20416 2007-05-15 16:46:10 +0000 Wim Taymans <wim.taymans@gmail.com>
20418 ext/ogg/gstoggdemux.c: Add some FIXMEs.
20419 Original commit message from CVS:
20420 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_packet),
20421 (gst_ogg_demux_activate_chain), (gst_ogg_demux_perform_seek),
20422 (gst_ogg_demux_handle_page):
20424 Fix chain start/stop segment handling based on patch by
20425 <ahalda at cs dot mcgill dot ca> see #320984.
20427 2007-05-15 15:33:54 +0000 Michael Smith <msmith@xiph.org>
20429 configure.ac: We don't require a C++ compiler. So don't require one.
20430 Original commit message from CVS:
20432 We don't require a C++ compiler. So don't require one.
20434 2007-05-15 15:29:17 +0000 Stefan Kost <ensonic@users.sourceforge.net>
20437 * ext/alsa/gstalsamixer.c:
20438 ext/alsa/gstalsamixer.c (source, n_poll_fds, poll_fds, gst_alsa_mixer_check, gst_alsa_mixer_dispatch, gst_alsa_mixer_...
20439 Original commit message from CVS:
20440 * ext/alsa/gstalsamixer.c (source, n_poll_fds, poll_fds,
20441 gst_alsa_mixer_check, gst_alsa_mixer_dispatch,
20442 gst_alsa_mixer_finalize, gst_alsa_mixer_handle_source_callback,
20443 gst_alsa_mixer_handle_callback, gst_alsa_mixer_elem_handle_callback,
20444 gst_alsa_mixer_ensure_track_list, gst_alsa_mixer_update_option,
20445 gst_alsa_mixer_update_track):
20446 Apply some of the cleanup Tim suggested in #152864 afterwards.
20448 2007-05-15 14:01:26 +0000 Marc-Andre Lureau <marcandre.lureau@gmail.com>
20450 ext/alsa/gstalsamixer.c (main_context, GstAlsaMixerWatch, _GstAlsaMixerWatch, source, n_poll_fds, poll_fds, gst_alsa_...
20451 Original commit message from CVS:
20452 patch by: Marc-Andre Lureau <marcandre.lureau@gmail.com>
20453 * ext/alsa/gstalsamixer.c (main_context, GstAlsaMixerWatch,
20454 _GstAlsaMixerWatch, source, n_poll_fds, poll_fds,
20455 gst_alsa_mixer_watch_funcs, gst_alsa_mixer_prepare,
20456 gst_alsa_mixer_check, gst_alsa_mixer_dispatch,
20457 gst_alsa_mixer_finalize, gst_alsa_mixer_find_master_mixer,
20458 gst_alsa_mixer_handle_source_callback,
20459 gst_alsa_mixer_handle_callback, gst_alsa_mixer_elem_handle_callback,
20460 gst_alsa_mixer_ensure_track_list, gst_alsa_mixer_free,
20461 gst_alsa_mixer_get_volume, gst_alsa_mixer_set_volume,
20462 gst_alsa_mixer_set_mute, gst_alsa_mixer_set_record,
20463 gst_alsa_mixer_get_option, gst_alsa_mixer_update_option,
20464 gst_alsa_mixer_update_track, _gst_alsa_mixer_set_interface):
20465 * ext/alsa/gstalsamixer.h (handle_source, interface, dir):
20466 * ext/alsa/gstalsamixerelement.c (gst_alsa_mixer_element_details,
20467 gst_alsa_mixer_element_interface_supported,
20468 gst_alsa_mixer_element_finalize, gst_alsa_mixer_element_init,
20469 gst_alsa_mixer_element_set_property,
20470 gst_alsa_mixer_element_get_property,
20471 gst_alsa_mixer_element_change_state):
20472 * ext/alsa/gstalsamixertrack.c (gst_alsa_mixer_track_update):
20473 * gst-libs/gst/interfaces/mixer.c (gst_mixer_volume_changed,
20474 gst_mixer_option_changed):
20475 * gst-libs/gst/interfaces/mixer.h (set_option, get_option,
20476 volume_changed, option_changed, _gst_reserved):
20477 Implement notification for alsamixer. Fixes #152864
20479 2007-05-15 03:53:11 +0000 David Schleef <ds@schleef.org>
20481 gst/videotestsrc/videotestsrc.*: Add support for video/x-raw-bayer.
20482 Original commit message from CVS:
20483 * gst/videotestsrc/videotestsrc.c:
20484 * gst/videotestsrc/videotestsrc.h:
20485 Add support for video/x-raw-bayer.
20487 2007-05-13 01:06:19 +0000 David Schleef <ds@schleef.org>
20489 sys/xvimage/xvimagesink.c: Add some sanity checking for the XVImage size returned by X.
20490 Original commit message from CVS:
20491 * sys/xvimage/xvimagesink.c:
20492 Add some sanity checking for the XVImage size returned by X.
20493 Related to #377400.
20495 2007-05-12 16:18:39 +0000 Wim Taymans <wim.taymans@gmail.com>
20497 gst-libs/gst/rtp/gstbasertpdepayload.c: Parse and use additional caps fields as described in updated application/x-rt...
20498 Original commit message from CVS:
20499 * gst-libs/gst/rtp/gstbasertpdepayload.c:
20500 (gst_base_rtp_depayload_setcaps),
20501 (gst_base_rtp_depayload_set_gst_timestamp):
20502 Parse and use additional caps fields as described in updated
20503 application/x-rtp caps spec.
20505 2007-05-12 16:16:22 +0000 Wim Taymans <wim.taymans@gmail.com>
20507 ext/ogg/gstoggdemux.c: If there is a stream in a chain without any data packets, ignore the stream in the total lengt...
20508 Original commit message from CVS:
20509 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_packet),
20510 (gst_ogg_demux_collect_chain_info):
20511 If there is a stream in a chain without any data packets, ignore the
20512 stream in the total length calculations. Might be related to #436820.
20514 2007-05-11 17:33:43 +0000 Jan Schmidt <thaytan@mad.scientist.com>
20516 gst/typefind/gsttypefindfunctions.c: Consolidate and re-work our mpeg system stream detection to probe more packets a...
20517 Original commit message from CVS:
20518 * gst/typefind/gsttypefindfunctions.c: (mpeg_sys_is_valid_pack),
20519 (mpeg_sys_is_valid_pes), (mpeg_sys_is_valid_sys),
20520 (mpeg_sys_type_find), (mpeg_ts_type_find), (mpeg4_video_type_find),
20521 (mpeg_video_type_find), (mpeg_video_stream_type_find),
20523 Consolidate and re-work our mpeg system stream detection to probe
20524 more packets and produce a higher confidence result. Fixes a
20525 regression caused by lowering the typefind probability last year
20526 - related to bug #397810. Remove the redundant MPEG-1 specific
20527 typefind function, as the new one detects both MPEG-1 & MPEG-2
20529 Also cleanup the MPEG elementary and MPEG-TS detection functions a
20531 Tested against my media test directory, with some improvements and
20534 2007-05-10 15:28:13 +0000 Wim Taymans <wim.taymans@gmail.com>
20536 gst/playback/gstplaybasebin.c: Connect to the new queue "pushing" signal instead of the broken "running" one.
20537 Original commit message from CVS:
20538 * gst/playback/gstplaybasebin.c: (fill_buffer), (check_queue),
20539 (queue_out_of_data):
20540 Connect to the new queue "pushing" signal instead of the broken
20543 2007-05-09 21:17:40 +0000 Sébastien Moutte <sebastien@moutte.net>
20545 gst-libs/gst/rtp/gstbasertpaudiopayload.c: Move variable declaration before the first instruction.
20546 Original commit message from CVS:
20547 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
20548 (gst_base_rtp_audio_payload_handle_frame_based_buffer):
20549 Move variable declaration before the first instruction.
20550 * gst/videotestsrc/videotestsrc.c:
20551 Define M_PI if it's not defined yet.
20552 * win32/common/libgstrtp.def:
20553 Add new exported functions.
20555 2007-05-09 11:54:32 +0000 Michael Smith <msmith@xiph.org>
20557 ext/theora/theoradec.c: gst_pad_push_event() does not return a GstFlowReturn!
20558 Original commit message from CVS:
20559 * ext/theora/theoradec.c: (theora_handle_type_packet):
20560 gst_pad_push_event() does not return a GstFlowReturn!
20562 2007-05-09 11:25:34 +0000 Wim Taymans <wim.taymans@gmail.com>
20564 tests/examples/seek/: Some small cosmetic changes.
20565 Original commit message from CVS:
20566 * tests/examples/seek/scrubby.c: (stop_cb), (main):
20567 * tests/examples/seek/seek.c: (do_seek):
20568 Some small cosmetic changes.
20570 2007-05-08 19:24:01 +0000 Stefan Kost <ensonic@users.sourceforge.net>
20573 * gst/adder/gstadder.c:
20574 * gst/adder/gstadder.h:
20575 gst/adder/gstadder.c (gst_adder_src_event, gst_adder_collected, gst_adder_change_state): gst/adder/gstadder.h (bps, o...
20576 Original commit message from CVS:
20577 * gst/adder/gstadder.c (gst_adder_src_event, gst_adder_collected,
20578 gst_adder_change_state):
20579 * gst/adder/gstadder.h (bps, offset, collect_event, segment,
20580 segment_pending, segment_position, segment_rate):
20581 Handle playback-rate on adder.
20583 2007-05-07 11:43:31 +0000 Michael Smith <msmith@xiph.org>
20585 ext/theora/: Don't push events (newsegment, tags) before initialising the decoder.
20586 Original commit message from CVS:
20587 * ext/theora/gsttheoradec.h:
20588 * ext/theora/theoradec.c: (gst_theora_dec_reset),
20589 (theora_dec_sink_event), (theora_handle_comment_packet),
20590 (theora_handle_type_packet), (theora_dec_change_state):
20591 Don't push events (newsegment, tags) before initialising the
20593 This is neccesary for seeking to work correctly in gnonlin.
20595 2007-05-04 13:10:07 +0000 Stefan Kost <ensonic@users.sourceforge.net>
20597 gst/: gst/audiotestsrc/gstaudiotestsrc.c
20598 Original commit message from CVS:
20599 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
20600 * gst/adder/gstadder.c:
20601 * gst/audiotestsrc/gstaudiotestsrc.c
20602 (gst_audio_test_src_create_white_noise):
20603 * gst/videotestsrc/gstvideotestsrc.c:
20604 * gst/volume/gstvolume.c (VOLUME_UNITY_INT16,
20605 VOLUME_UNITY_INT16_BIT_SHIFT, VOLUME_MAX_DOUBLE,
20606 volume_sink_template, volume_src_template, gst_volume_init,
20607 volume_process_double, volume_process_int16,
20608 volume_process_int16_clamp):
20609 Doc fixes and formatting.
20611 2007-05-04 12:41:21 +0000 Tim-Philipp Müller <tim@centricular.net>
20613 tests/check/: Minimal check for volume's GstController usability; also another test for #422295.
20614 Original commit message from CVS:
20615 * tests/check/Makefile.am:
20616 * tests/check/elements/volume.c: (GST_START_TEST), (volume_suite):
20617 Minimal check for volume's GstController usability; also another
20620 2007-05-04 09:06:38 +0000 Tim-Philipp Müller <tim@centricular.net>
20622 gst-libs/gst/cdda/gstcddabasesrc.c: Fix it so that it (a) makes sense and (b) doesn't break everything cdda-related i...
20623 Original commit message from CVS:
20624 * gst-libs/gst/cdda/gstcddabasesrc.c:
20625 (gst_cdda_base_src_add_track):
20626 Fix it so that it (a) makes sense and (b) doesn't break
20627 everything cdda-related including the unit test.
20629 2007-05-04 08:46:59 +0000 Stefan Kost <ensonic@users.sourceforge.net>
20631 gst-libs/gst/cdda/gstcddabasesrc.c: Fix build when disabling asserts.
20632 Original commit message from CVS:
20633 * gst-libs/gst/cdda/gstcddabasesrc.c:
20634 (gst_cdda_base_src_add_track):
20635 Fix build when disabling asserts.
20637 2007-05-03 16:29:10 +0000 Tim-Philipp Müller <tim@centricular.net>
20639 sys/ximage/ximagesink.c: When XShm is not available, we might get row strides that are not rounded up to multiples of...
20640 Original commit message from CVS:
20641 * sys/ximage/ximagesink.c: (gst_ximagesink_ximage_new):
20642 When XShm is not available, we might get row strides that are not
20643 rounded up to multiples of four; this is bad, because virtually
20644 every RGB-processing element in GStreamer assumes rowstrides are
20645 rounded up to multiples of four, so let's allocate at least enough
20646 memory to avoid crashes in this case. The image will still be
20647 displayed distorted though if this happens, so that still needs
20648 fixing (maybe by allocating a bigger image with an 'even' width
20649 and then clipping it appropriately when rendering - something for
20650 Xlib aficionados in any case).
20652 2007-05-03 13:16:21 +0000 Michael Smith <msmith@xiph.org>
20654 gst/audiorate/gstaudiorate.c: If a buffer doesn't have a timestamp, assume it's contiguous with the previous buffer, ...
20655 Original commit message from CVS:
20656 * gst/audiorate/gstaudiorate.c: (gst_audio_rate_chain):
20657 If a buffer doesn't have a timestamp, assume it's contiguous with
20658 the previous buffer, and synthesise timestamps appropriately.
20660 2007-05-03 11:24:00 +0000 Edward Hervey <bilboed@bilboed.com>
20662 tests/check/elements/videorate.c: Set buffer timestamp to a valid value in order to test the buffer really does stay ...
20663 Original commit message from CVS:
20664 * tests/check/elements/videorate.c: (GST_START_TEST):
20665 Set buffer timestamp to a valid value in order to test the buffer
20666 really does stay in videorate.
20668 2007-05-03 10:47:22 +0000 Edward Hervey <bilboed@bilboed.com>
20670 gst/videorate/gstvideorate.c: There is no sensible way to handle incoming buffers which don't have a valid timestamp....
20671 Original commit message from CVS:
20672 * gst/videorate/gstvideorate.c: (gst_video_rate_chain):
20673 There is no sensible way to handle incoming buffers which don't have a
20674 valid timestamp. We therefore discard them and wait for the next one.
20676 2007-05-01 18:45:36 +0000 Tim-Philipp Müller <tim@centricular.net>
20678 gst/playback/: Better error message for text files.
20679 Original commit message from CVS:
20680 * gst/playback/gstdecodebin.c: (type_found), (plugin_init):
20681 * gst/playback/gstdecodebin2.c: (plugin_init):
20682 Better error message for text files.
20684 2007-04-29 14:38:05 +0000 Wim Taymans <wim.taymans@gmail.com>
20686 gst-libs/gst/rtp/gstrtcpbuffer.c: Fix offset bug in generation RR packets.
20687 Original commit message from CVS:
20688 * gst-libs/gst/rtp/gstrtcpbuffer.c: (gst_rtcp_packet_add_rb):
20689 Fix offset bug in generation RR packets.
20691 2007-04-27 15:33:46 +0000 Julien Moutte <julien@moutte.net>
20693 ext/theora/theoradec.c: Calculate buffer duration correctly to generate a perfect stream (#433888).
20694 Original commit message from CVS:
20695 2007-04-27 Julien MOUTTE <julien@moutte.net>
20696 * ext/theora/theoradec.c: (_theora_granule_time),
20697 (theora_dec_push_forward), (theora_handle_data_packet),
20698 (theora_dec_decode_buffer): Calculate buffer duration correctly
20699 to generate a perfect stream (#433888).
20700 * gst/audioresample/gstaudioresample.c:
20701 (audioresample_check_discont): Glib provides ABS.
20703 2007-04-27 15:01:40 +0000 Wim Taymans <wim.taymans@gmail.com>
20705 gst-libs/gst/rtp/gstrtcpbuffer.*: Fix RB block parsing and writing.
20706 Original commit message from CVS:
20707 * gst-libs/gst/rtp/gstrtcpbuffer.c: (gst_rtcp_packet_get_rb),
20708 (gst_rtcp_packet_add_rb), (gst_rtcp_packet_sdes_add_item),
20709 (gst_rtcp_packet_sdes_add_entry), (gst_rtcp_packet_bye_add_ssrc),
20710 (gst_rtcp_packet_bye_add_ssrcs), (get_reason_offset),
20711 (gst_rtcp_packet_bye_set_reason):
20712 * gst-libs/gst/rtp/gstrtcpbuffer.h:
20713 Fix RB block parsing and writing.
20714 Add support for constructing BYE packets.
20716 2007-04-25 08:54:34 +0000 Tim-Philipp Müller <tim@centricular.net>
20718 When posting a warning message because samples were dropped, post something more intelligible than he default error m...
20719 Original commit message from CVS:
20720 * gst-libs/gst/audio/gstbaseaudiosrc.c: (_do_init),
20721 (gst_base_audio_src_create):
20723 When posting a warning message because samples were dropped, post
20724 something more intelligible than he default error message for clock
20725 errors which is just confusing in this context (#432984).
20727 2007-04-25 08:10:26 +0000 Wim Taymans <wim.taymans@gmail.com>
20729 gst-libs/gst/rtp/gstrtcpbuffer.*: Implement code to write SR, RR and SDES packets.
20730 Original commit message from CVS:
20731 * gst-libs/gst/rtp/gstrtcpbuffer.c: (gst_rtcp_buffer_new),
20732 (gst_rtcp_buffer_end), (gst_rtcp_buffer_get_packet_count),
20733 (read_packet_header), (gst_rtcp_packet_move_to_next),
20734 (gst_rtcp_buffer_add_packet), (gst_rtcp_packet_sr_set_sender_info),
20735 (gst_rtcp_packet_rr_set_ssrc), (gst_rtcp_packet_add_rb),
20736 (gst_rtcp_packet_sdes_get_item_count),
20737 (gst_rtcp_packet_sdes_first_item),
20738 (gst_rtcp_packet_sdes_next_item), (gst_rtcp_packet_sdes_get_ssrc),
20739 (gst_rtcp_packet_sdes_first_entry),
20740 (gst_rtcp_packet_sdes_next_entry),
20741 (gst_rtcp_packet_sdes_get_entry), (gst_rtcp_packet_sdes_add_item),
20742 (gst_rtcp_packet_sdes_add_entry):
20743 * gst-libs/gst/rtp/gstrtcpbuffer.h:
20744 Implement code to write SR, RR and SDES packets.
20746 2007-04-24 20:45:24 +0000 Christian Kirbach <Christian.Kirbach@googlemail.com>
20748 sys/ximage/ximagesink.c: Fix build if XShm is not available (#432362).
20749 Original commit message from CVS:
20750 Patch by: Christian Kirbach <Christian dot Kirbach at googlemail com>
20751 * sys/ximage/ximagesink.c:
20752 Fix build if XShm is not available (#432362).
20754 2007-04-24 18:58:25 +0000 Sebastian Dröge <slomo@circular-chaos.org>
20756 gst/audioconvert/gstaudioconvert.c: Initalize the AudioConvertCtx with zeroes, otherwise it will contain pointers to ...
20757 Original commit message from CVS:
20758 * gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_init):
20759 Initalize the AudioConvertCtx with zeroes, otherwise it will contain
20760 pointers to random memory which are passed to g_free() when
20761 audio_convert_prepare_context() is called the first time.
20763 2007-04-24 15:00:07 +0000 Dan Williams <dcbw@redhat.com>
20765 gst/videorate/gstvideorate.c: Don't leak incoming buffer if gst_pad_push() returns a non-OK flow. Fixes #432755.
20766 Original commit message from CVS:
20767 Patch by: Dan Williams <dcbw redhat com>
20768 * gst/videorate/gstvideorate.c: (gst_video_rate_chain):
20769 Don't leak incoming buffer if gst_pad_push() returns a
20770 non-OK flow. Fixes #432755.
20771 * tests/check/elements/videorate.c: (GST_START_TEST),
20773 Unit test for the above by Yours Truly.
20775 2007-04-23 20:04:28 +0000 Stefan Kost <ensonic@users.sourceforge.net>
20777 gst/adder/gstadder.c: Fix non-flushing segmented seeks, Fixes #340060 for me
20778 Original commit message from CVS:
20779 * gst/adder/gstadder.c: (gst_adder_setcaps), (gst_adder_src_event),
20780 (gst_adder_sink_event), (gst_adder_collected):
20781 Fix non-flushing segmented seeks, Fixes #340060 for me
20783 2007-04-21 15:29:27 +0000 Tim-Philipp Müller <tim@centricular.net>
20786 ChangeLog surgery: add API keyword
20787 Original commit message from CVS:
20788 ChangeLog surgery: add API keyword
20790 2007-04-21 15:25:22 +0000 Olivier Crete <tester@tester.ca>
20792 gst-libs/gst/rtp/gstbasertpaudiopayload.c: Chain up to parent class in dispose function; get rid of unnecessary 'dipo...
20793 Original commit message from CVS:
20794 Patch by: Olivier Crete <tester at tester ca>
20795 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
20796 (gst_base_rtp_audio_payload_class_init),
20797 (gst_base_rtp_audio_payload_init),
20798 (gst_base_rtp_audio_payload_dispose):
20799 Chain up to parent class in dispose function; get rid of
20800 unnecessary 'diposed' flag in private structure (#415001).
20802 2007-04-21 15:10:25 +0000 Tim-Philipp Müller <tim@centricular.net>
20804 Some minor docs fixes and additions; also add missing 'Since' bits.
20805 Original commit message from CVS:
20806 * docs/libs/gst-plugins-base-libs.types:
20807 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
20808 (gst_base_rtp_audio_payload_class_init):
20809 * gst-libs/gst/rtp/gstbasertpdepayload.c:
20810 * gst-libs/gst/rtp/gstbasertppayload.c:
20811 Some minor docs fixes and additions; also add missing 'Since' bits.
20813 2007-04-21 14:40:45 +0000 Zeeshan Ali <zeenix@gmail.com>
20815 gst-libs/gst/rtp/gstbasertpaudiopayload.*: The recently-added gst_base_rtp_audio_payload_push() should take an object...
20816 Original commit message from CVS:
20817 Patch by: Zeeshan Ali <zeenix gmail com>
20818 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
20819 (gst_base_rtp_audio_payload_handle_frame_based_buffer),
20820 (gst_base_rtp_audio_payload_handle_sample_based_buffer),
20821 (gst_base_rtp_audio_payload_push):
20822 * gst-libs/gst/rtp/gstbasertpaudiopayload.h:
20823 The recently-added gst_base_rtp_audio_payload_push() should take an
20824 object of type GstBaseRTPAudioPayload as first argument (#431672).
20826 2007-04-21 14:14:24 +0000 Tim-Philipp Müller <tim@centricular.net>
20828 gst/audioresample/gstaudioresample.c: Make more functions static, just because we can.
20829 Original commit message from CVS:
20830 * gst/audioresample/gstaudioresample.c:
20831 Make more functions static, just because we can.
20833 2007-04-21 13:54:39 +0000 Tim-Philipp Müller <tim@centricular.net>
20835 tests/check/elements/audioresample.c: Add unit test for audioresample shutdown crasher (#420106).
20836 Original commit message from CVS:
20837 * tests/check/elements/audioresample.c:
20838 Add unit test for audioresample shutdown crasher (#420106).
20840 2007-04-20 10:42:24 +0000 Stefan Kost <ensonic@users.sourceforge.net>
20842 gst/subparse/: Use GST_DISABLE_XML here
20843 Original commit message from CVS:
20844 * gst/subparse/gstsubparse.c:
20845 * gst/subparse/samiparse.c:
20846 Use GST_DISABLE_XML here
20847 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_put),
20848 (gst_xvimagesink_xwindow_new), (gst_xvimagesink_get_xv_support),
20849 (gst_xvimagesink_buffer_alloc),
20850 (gst_xvimagesink_navigation_send_event):
20851 * sys/xvimage/xvimagesink.h:
20852 Include stdlib.h when using atoi.
20853 * tests/check/elements/playbin.c: (playbin_suite):
20854 Use GST_DISABLE_REGISTRY here
20856 2007-04-19 16:58:53 +0000 Michael Smith <msmith@xiph.org>
20858 ext/theora/: Track initialisation state; don't try to use encoder state if we're not initialised (it'll segfault).
20859 Original commit message from CVS:
20860 * ext/theora/gsttheoraenc.h:
20861 * ext/theora/theoraenc.c: (theora_enc_sink_setcaps),
20862 (theora_enc_sink_event), (theora_enc_change_state):
20863 Track initialisation state; don't try to use encoder state if we're
20864 not initialised (it'll segfault).
20866 2007-04-18 11:06:42 +0000 Stefan Kost <ensonic@users.sourceforge.net>
20868 tests/check/pipelines/.cvsignore: Fix build.
20869 Original commit message from CVS:
20870 * tests/check/pipelines/.cvsignore:
20873 2007-04-17 10:56:37 +0000 Tim-Philipp Müller <tim@centricular.net>
20875 gst/app/Makefile.am: Fix CFLAGS and hopefully #430594.
20876 Original commit message from CVS:
20877 * gst/app/Makefile.am:
20878 Fix CFLAGS and hopefully #430594.
20880 2007-04-17 02:53:16 +0000 Sebastian Dröge <slomo@circular-chaos.org>
20882 gst-libs/gst/riff/riff-media.c: Allow random depths between 1 and 32 instead of only multiplies of 8.
20883 Original commit message from CVS:
20884 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
20885 Allow random depths between 1 and 32 instead of only multiplies of 8.
20887 2007-04-17 02:04:21 +0000 Sebastian Dröge <slomo@circular-chaos.org>
20889 gst-libs/gst/riff/riff-media.c: Set the maximum number of channels for PCM and float in the correct place to have it ...
20890 Original commit message from CVS:
20891 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
20892 Set the maximum number of channels for PCM and float in the correct
20893 place to have it also used when creating the template caps.
20895 2007-04-17 01:56:07 +0000 Sebastian Dröge <slomo@circular-chaos.org>
20897 gst-libs/gst/riff/riff-media.c: Correctly support 4, 6 and 8 channels with normal PCM and float wav files.
20898 Original commit message from CVS:
20899 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
20900 Correctly support 4, 6 and 8 channels with normal PCM and float
20902 Fix the depth and signedness calculation in extensible wav files and
20903 also handle 1, 2, 4, 6, 8 channels here when a file without channel
20905 Add support for float, alaw and mulaw in extensible wav files.
20906 This allows correct playback of all but 5 files from
20907 http://www-mmsp.ece.mcgill.ca/Documents/AudioFormats/WAVE/Samples.html
20908 (gst_riff_create_audio_template_caps):
20909 Add voxware and float formats to the template caps.
20911 2007-04-16 22:20:03 +0000 Vincent Torri <vtorri@univ-evry.fr>
20913 ext/pango/gstclockoverlay.c: Fix unused variable warning if HAVE_LOCALTIME_R is undefinied
20914 Original commit message from CVS:
20915 Patch by: Vincent Torri <vtorri at univ-evry dot fr>
20916 * ext/pango/gstclockoverlay.c: (gst_clock_overlay_render_time):
20917 Fix unused variable warning if HAVE_LOCALTIME_R is undefinied
20918 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
20919 * gst/audioresample/gstaudioresample.c: (audioresample_do_output):
20920 Use the correct format strings for integer formats.
20922 2007-04-16 21:44:34 +0000 Thomas Vander Stichele <thomas@apestaart.org>
20924 * gst-plugins-base.doap:
20926 Original commit message from CVS:
20929 2007-04-16 21:42:13 +0000 Thomas Vander Stichele <thomas@apestaart.org>
20931 * gst-plugins-base.doap:
20933 Original commit message from CVS:
20936 2007-04-15 14:35:53 +0000 Thomas Vander Stichele <thomas@apestaart.org>
20938 ext/theora/theoraenc.c (theora_buffer_from_packet, theora_enc_chain): Don't use pad_alloc_buffer_and_set_caps to crea...
20939 Original commit message from CVS:
20940 * ext/theora/theoraenc.c (theora_buffer_from_packet, theora_enc_chain):
20941 Don't use pad_alloc_buffer_and_set_caps to create a small header
20942 packet, or, worse, to create a big temporary video buffer using the
20945 2007-04-14 12:34:55 +0000 Thomas Vander Stichele <thomas@apestaart.org>
20947 gst/gdp/gstgdppay.c (gst_gdp_pay_chain): tests/check/pipelines/streamheader.c (tag_event_probe_cb,
20948 Original commit message from CVS:
20949 * gst/gdp/gstgdppay.c (gst_gdp_pay_chain):
20950 * tests/check/pipelines/streamheader.c (tag_event_probe_cb,
20951 GST_START_TEST, buffer_probe_cb, GST_START_TEST):
20952 Fix a bug where serialized IN_CAPS buffers needed to be set IN_CAPS.
20954 2007-04-13 22:10:58 +0000 Thomas Vander Stichele <thomas@apestaart.org>
20956 * gst/tcp/gstmultifdsink.c:
20958 Original commit message from CVS:
20961 2007-04-13 21:55:31 +0000 Thomas Vander Stichele <thomas@apestaart.org>
20964 * tests/check/pipelines/streamheader.c:
20965 tests/check/pipelines/streamheader.c (tag_event_probe_cb,
20966 Original commit message from CVS:
20967 * tests/check/pipelines/streamheader.c (tag_event_probe_cb,
20968 GST_START_TEST, n_in_caps, buffer_probe_cb, GST_START_TEST,
20969 streamheader_suite):
20970 Add another test set up for failure
20972 2007-04-13 21:09:04 +0000 Thomas Vander Stichele <thomas@apestaart.org>
20974 * ext/ogg/gstoggmux.c:
20975 * gst/gdp/gstgdpdepay.c:
20977 Original commit message from CVS:
20980 2007-04-13 21:08:11 +0000 Thomas Vander Stichele <thomas@apestaart.org>
20982 tests/check/Makefile.am: tests/check/pipelines/streamheader.c (n_tags, tag_event_probe_cb,
20983 Original commit message from CVS:
20984 * tests/check/Makefile.am:
20985 * tests/check/pipelines/streamheader.c (n_tags, tag_event_probe_cb,
20986 GST_START_TEST, streamheader_suite, main):
20987 Add a test for the streamheader bug Wim fixed.
20989 2007-04-13 11:42:34 +0000 Jan Schmidt <thaytan@mad.scientist.com>
20991 ext/theora/theoradec.c: Fix misleading comment.
20992 Original commit message from CVS:
20993 * ext/theora/theoradec.c: (theora_dec_sink_event):
20994 Fix misleading comment.
20996 2007-04-13 06:17:45 +0000 Stefan Kost <ensonic@users.sourceforge.net>
20998 gst-libs/gst/riff/riff-media.c: More sanity checks for the header fields.
20999 Original commit message from CVS:
21000 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
21001 More sanity checks for the header fields.
21003 2007-04-12 16:36:36 +0000 Tim-Philipp Müller <tim@centricular.net>
21005 gst-libs/gst/tag/tags.c: Try encodings from all environment variables, not just those in the first environment variab...
21006 Original commit message from CVS:
21007 * gst-libs/gst/tag/tags.c: (gst_tag_freeform_string_to_utf8):
21008 Try encodings from all environment variables, not just those in the
21009 first environment variable that is set.
21011 2007-04-12 15:00:03 +0000 Wim Taymans <wim.taymans@gmail.com>
21013 gst/videorate/gstvideorate.c: Add some debug.
21014 Original commit message from CVS:
21015 * gst/videorate/gstvideorate.c: (gst_video_rate_setcaps),
21016 (gst_video_rate_chain):
21018 * tests/check/elements/videorate.c: (GST_START_TEST),
21020 Added check for videorate changing caps handling. Closes #421834.
21022 2007-04-12 12:57:33 +0000 Michael Smith <msmith@xiph.org>
21024 ext/vorbis/vorbisdec.c: Use scale functions to avoid overflow when calculating duration of vorbis buffers.
21025 Original commit message from CVS:
21026 * ext/vorbis/vorbisdec.c: (vorbis_handle_data_packet):
21027 Use scale functions to avoid overflow when calculating duration of
21030 2007-04-12 12:19:20 +0000 Tim-Philipp Müller <tim@centricular.net>
21032 API: add gst_tag_freeform_string_to_utf8() (#405072).
21033 Original commit message from CVS:
21034 * docs/libs/gst-plugins-base-libs-sections.txt:
21035 * gst-libs/gst/tag/tag.h:
21036 * gst-libs/gst/tag/tags.c: (gst_tag_freeform_string_to_utf8):
21037 API: add gst_tag_freeform_string_to_utf8() (#405072).
21038 * gst-libs/gst/tag/gstid3tag.c: (gst_tag_extract_id3v1_string):
21039 Use gst_tag_freeform_string_to_utf8() here.
21041 2007-04-12 10:38:03 +0000 Thomas Vander Stichele <thomas@apestaart.org>
21043 * gst/tcp/gstmultifdsink.c:
21045 Original commit message from CVS:
21048 2007-04-12 10:03:22 +0000 Wim Taymans <wim.taymans@gmail.com>
21050 gst/gdp/gstgdppay.c: Make sure we set the IN_CAPS flag correctly.
21051 Original commit message from CVS:
21052 * gst/gdp/gstgdppay.c: (gst_gdp_pay_chain),
21053 (gst_gdp_pay_sink_event):
21054 Make sure we set the IN_CAPS flag correctly.
21055 * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_render):
21056 Get the IN_CAPS flag before we call functions that mess with the flags.
21058 2007-04-10 20:37:05 +0000 Thomas Vander Stichele <thomas@apestaart.org>
21061 * gst/gdp/gstgdppay.c:
21062 gst/gdp/gstgdppay.c (gst_gdp_pay_reset_streamheader, gst_gdp_pay_chain, gst_gdp_pay_sink_event):
21063 Original commit message from CVS:
21064 * gst/gdp/gstgdppay.c (gst_gdp_pay_reset_streamheader,
21065 gst_gdp_pay_chain, gst_gdp_pay_sink_event):
21066 Only stamp buffers with offset/offset_end right before they get
21067 pushed. This ensures offset continuity, which was not the case
21069 gst-launch -v -m audiotestsrc num-buffers=10 ! audioconvert ! vorbisenc ! gdppay ! identity check-imperfect-offset=TRUE ! fakesink silent=TRUE
21071 2007-04-10 20:25:06 +0000 Thomas Vander Stichele <thomas@apestaart.org>
21073 * gst/gdp/gstgdpdepay.c:
21074 * gst/gdp/gstgdppay.c:
21076 Original commit message from CVS:
21079 2007-04-10 11:23:18 +0000 Christian Schaller <uraeus@gnome.org>
21082 * gst-plugins-base.spec.in:
21083 update spec file for RTP changes
21084 Original commit message from CVS:
21085 update spec file for RTP changes
21087 2007-04-06 12:58:06 +0000 Wim Taymans <wim.taymans@gmail.com>
21089 gst/playback/gstplaybin.c: Activate sync in playbin, we are ready to handle it for live streams.
21090 Original commit message from CVS:
21091 * gst/playback/gstplaybin.c: (add_sink),
21092 (gst_play_bin_change_state):
21093 Activate sync in playbin, we are ready to handle it for live streams.
21095 2007-04-06 09:56:18 +0000 Tim-Philipp Müller <tim@centricular.net>
21097 tests/check/elements/playbin.c: Add small test for stream-info-value-array code paths.
21098 Original commit message from CVS:
21099 * tests/check/elements/playbin.c:
21100 (test_sink_usage_video_only_stream), (playbin_suite):
21101 Add small test for stream-info-value-array code paths.
21103 2007-04-05 15:44:40 +0000 Wim Taymans <wim.taymans@gmail.com>
21105 gst-libs/gst/audio/gstbaseaudiosink.c: Don't try to create invalid calibration parameters by making the internal time...
21106 Original commit message from CVS:
21107 * gst-libs/gst/audio/gstbaseaudiosink.c:
21108 (gst_base_audio_sink_skew_slaving):
21109 Don't try to create invalid calibration parameters by making the
21110 internal time go backwards, instead make external time go forward.
21112 2007-04-05 10:27:06 +0000 Tommi Myöhänen <ext-tommi.myohanen@nokia.com>
21114 gst/playback/gstplaybasebin.c: Fix leak in add_stream(), when g_value_set_object() increases the refcount of streamin...
21115 Original commit message from CVS:
21116 Patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com>
21117 * gst/playback/gstplaybasebin.c: (add_stream):
21118 Fix leak in add_stream(), when g_value_set_object() increases the
21119 refcount of streaminfo object. Fixes #426250.
21121 2007-04-04 02:45:03 +0000 David Schleef <ds@schleef.org>
21123 gst/videotestsrc/: Add a test pattern called "circular", which has concentric rings with varying radial frequency. T...
21124 Original commit message from CVS:
21125 * gst/videotestsrc/gstvideotestsrc.c:
21126 * gst/videotestsrc/gstvideotestsrc.h:
21127 * gst/videotestsrc/videotestsrc.c:
21128 * gst/videotestsrc/videotestsrc.h:
21129 Add a test pattern called "circular", which has concentric
21130 rings with varying radial frequency. The main purpose of this
21131 pattern is to test fidelity loss in a filter or scaler element.
21132 Notably, this pattern is scale invariant, and is optimally viewed
21133 with a width (and height) of 400.
21135 2007-04-03 11:10:52 +0000 Tommi Myöhänen <ext-tommi.myohanen@nokia.com>
21137 gst/playback/gstdecodebin2.c: Decodebin2 doesn't unref pads it obtains in some occasions:
21138 Original commit message from CVS:
21139 Patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com>
21140 * gst/playback/gstdecodebin2.c: (connect_pad), (expose_pad),
21141 (deactivate_free_recursive):
21142 Decodebin2 doesn't unref pads it obtains in some occasions:
21143 - multiqueue src pads, when either connecting further or exposing
21144 - sink pads of new autoplugged elements
21145 - peer pads when recursively freeing elements
21148 2007-03-30 17:05:23 +0000 Sebastian Dröge <slomo@circular-chaos.org>
21150 gst-libs/gst/riff/riff-media.c: Add audio/x-raw-float support, now that audioconvert support non-native endianness fl...
21151 Original commit message from CVS:
21152 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
21153 Add audio/x-raw-float support, now that audioconvert support
21154 non-native endianness floats.
21156 2007-03-30 15:00:49 +0000 Tim-Philipp Müller <tim@centricular.net>
21158 docs/libs/gst-plugins-base-libs-docs.sgml: gstreamer-plugins-base.pc doesn't exist, it's gstreamer-plugins-base-0.10.pc.
21159 Original commit message from CVS:
21160 * docs/libs/gst-plugins-base-libs-docs.sgml:
21161 gstreamer-plugins-base.pc doesn't exist, it's
21162 gstreamer-plugins-base-0.10.pc.
21164 2007-03-29 18:42:34 +0000 René Stadler <mail@renestadler.de>
21166 with some minor changes
21167 Original commit message from CVS:
21168 Patch by: René Stadler <mail at renestadler dot de>
21169 with some minor changes
21170 * gst-libs/gst/floatcast/floatcast.h:
21171 Use more efficient float endianness conversion functions that don't
21172 involve 2 function calls per value.
21173 * gst/audioconvert/audioconvert.c: (audio_convert_get_func_index),
21174 (check_default), (audio_convert_prepare_context):
21175 * gst/audioconvert/gstaudioconvert.c:
21176 (gst_audio_convert_parse_caps), (make_lossless_changes):
21177 Support non-native endianness floats as input and output.
21179 * tests/check/elements/audioconvert.c: (verify_convert),
21181 Add unit tests for the non-native endianness float conversions.
21183 2007-03-29 16:23:53 +0000 Wim Taymans <wim.taymans@gmail.com>
21185 gst-libs/gst/rtp/gstbasertpdepayload.*: Add Private structure.
21186 Original commit message from CVS:
21187 * gst-libs/gst/rtp/gstbasertpdepayload.c:
21188 (gst_base_rtp_depayload_base_init),
21189 (gst_base_rtp_depayload_class_init), (gst_base_rtp_depayload_init),
21190 (gst_base_rtp_depayload_setcaps), (gst_base_rtp_depayload_chain),
21191 (gst_base_rtp_depayload_set_gst_timestamp),
21192 (gst_base_rtp_depayload_change_state),
21193 (gst_base_rtp_depayload_set_property),
21194 (gst_base_rtp_depayload_get_property):
21195 * gst-libs/gst/rtp/gstbasertpdepayload.h:
21196 Add Private structure.
21197 Bring element code to 2007.
21198 Parse clock-base caps param and use it when generating the
21200 Reset variables before going to PAUSED.
21203 2007-03-29 16:20:31 +0000 Wim Taymans <wim.taymans@gmail.com>
21206 Original commit message from CVS:
21207 * docs/libs/gst-plugins-base-libs-docs.sgml:
21208 * docs/libs/gst-plugins-base-libs-sections.txt:
21209 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
21210 (gst_base_rtp_audio_payload_get_adapter):
21212 Fix some more docs.
21213 * gst-libs/gst/rtp/Makefile.am:
21214 * gst-libs/gst/rtp/gstrtcpbuffer.c:
21215 (gst_rtcp_buffer_new_take_data), (gst_rtcp_buffer_new_copy_data),
21216 (gst_rtcp_buffer_validate_data), (gst_rtcp_buffer_validate),
21217 (gst_rtcp_buffer_get_packet_count), (read_packet_header),
21218 (gst_rtcp_buffer_get_first_packet), (gst_rtcp_packet_move_to_next),
21219 (gst_rtcp_buffer_add_packet), (gst_rtcp_packet_remove),
21220 (gst_rtcp_packet_get_padding), (gst_rtcp_packet_get_type),
21221 (gst_rtcp_packet_get_count), (gst_rtcp_packet_get_length),
21222 (gst_rtcp_packet_sr_get_sender_info),
21223 (gst_rtcp_packet_sr_set_sender_info),
21224 (gst_rtcp_packet_rr_get_ssrc), (gst_rtcp_packet_rr_set_ssrc),
21225 (gst_rtcp_packet_get_rb_count), (gst_rtcp_packet_get_rb),
21226 (gst_rtcp_packet_add_rb), (gst_rtcp_packet_set_rb),
21227 (gst_rtcp_packet_sdes_get_chunk_count),
21228 (gst_rtcp_packet_sdes_first_chunk),
21229 (gst_rtcp_packet_sdes_next_chunk), (gst_rtcp_packet_sdes_get_ssrc),
21230 (gst_rtcp_packet_sdes_first_item),
21231 (gst_rtcp_packet_sdes_next_item), (gst_rtcp_packet_sdes_get_item),
21232 (gst_rtcp_packet_bye_get_ssrc_count),
21233 (gst_rtcp_packet_bye_get_nth_ssrc), (gst_rtcp_packet_bye_add_ssrc),
21234 (gst_rtcp_packet_bye_add_ssrcs), (get_reason_offset),
21235 (gst_rtcp_packet_bye_get_reason_len),
21236 (gst_rtcp_packet_bye_get_reason), (gst_rtcp_packet_bye_set_reason):
21237 * gst-libs/gst/rtp/gstrtcpbuffer.h:
21238 Add new helper object for parsing and creating RTCP messages.
21240 2007-03-29 12:07:02 +0000 Sebastian Dröge <slomo@circular-chaos.org>
21242 gst-libs/gst/riff/riff-media.c: PCM samples with width=8 must be always unsigned, no matter what depth they have.
21243 Original commit message from CVS:
21244 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
21245 PCM samples with width=8 must be always unsigned, no matter what
21248 2007-03-29 11:24:47 +0000 Andy Wingo <wingo@pobox.com>
21250 gst/videorate/gstvideorate.c (gst_video_rate_flush_prev): Make perfect offsets also, not just timestamps.
21251 Original commit message from CVS:
21252 2007-03-29 Andy Wingo <wingo@pobox.com>
21253 * gst/videorate/gstvideorate.c (gst_video_rate_flush_prev): Make
21254 perfect offsets also, not just timestamps.
21255 * tests/check/elements/videorate.c (test_more): Test that given
21256 any incoming offsets, that videorate produces perfect offsets.
21258 2007-03-29 10:19:45 +0000 Wim Taymans <wim.taymans@gmail.com>
21260 gst-libs/gst/riff/riff-ids.h: Add some more RIFF formats.
21261 Original commit message from CVS:
21262 * gst-libs/gst/riff/riff-ids.h:
21263 Add some more RIFF formats.
21265 2007-03-29 10:17:52 +0000 Wim Taymans <wim.taymans@gmail.com>
21267 gst-libs/gst/rtp/gstrtpbuffer.*: Fix fixed payload names and docs.
21268 Original commit message from CVS:
21269 * gst-libs/gst/rtp/gstrtpbuffer.c:
21270 (gst_rtp_buffer_default_clock_rate):
21271 * gst-libs/gst/rtp/gstrtpbuffer.h:
21272 Fix fixed payload names and docs.
21273 Added method to get the default clock rates of fixed payload types.
21274 API: GstRTPBuffer::gst_rtp_buffer_default_clock_rate()
21276 2007-03-28 15:24:40 +0000 Zaheer Abbas Merali <zaheerabbas@merali.org>
21278 tests/check/pipelines/.cvsignore: Add new vorbisdec test to cvsignore.
21279 Original commit message from CVS:
21280 * tests/check/pipelines/.cvsignore:
21281 Add new vorbisdec test to cvsignore.
21283 2007-03-28 14:50:47 +0000 Wim Taymans <wim.taymans@gmail.com>
21285 gst-libs/gst/audio/gstbaseaudiosink.*: Store private stuff in GstBaseAudioSinkPrivate.
21286 Original commit message from CVS:
21287 * gst-libs/gst/audio/gstbaseaudiosink.c: (slave_method_get_type),
21288 (gst_base_audio_sink_class_init), (gst_base_audio_sink_init),
21289 (gst_base_audio_sink_query), (gst_base_audio_sink_get_time),
21290 (gst_base_audio_sink_set_property),
21291 (gst_base_audio_sink_get_property), (gst_base_audio_sink_event),
21292 (clock_convert_external), (gst_base_audio_sink_resample_slaving),
21293 (gst_base_audio_sink_skew_slaving),
21294 (gst_base_audio_sink_handle_slaving), (gst_base_audio_sink_render),
21295 (gst_base_audio_sink_async_play):
21296 * gst-libs/gst/audio/gstbaseaudiosink.h:
21297 Store private stuff in GstBaseAudioSinkPrivate.
21298 Add configurable clock slaving modes property.
21299 API:: GstBaseAudioSink::slave-method property
21300 Some more latency reporting tweaks.
21301 Added skew based clock slaving correction and make it the default until
21302 the resampling method is more robust.
21304 2007-03-27 12:44:14 +0000 Sebastian Dröge <slomo@circular-chaos.org>
21306 gst/audioconvert/audioconvert.c: Add docs to the integer pack functions and implement proper rounding. Before we had ...
21307 Original commit message from CVS:
21308 * gst/audioconvert/audioconvert.c:
21309 Add docs to the integer pack functions and implement proper
21310 rounding. Before we had rounding towards negative infinity, i.e.
21311 always the smaller number was taken. Now we use natural rounding,
21312 i.e. rounding to the nearest integer and to the one with the largest
21313 absolute value for X.5. The old rounding introduced some minor
21314 distortions. Fixes #420079
21315 * tests/check/elements/audioconvert.c: (GST_START_TEST):
21316 Fix one unit test that assumed the old rounding and added unit tests
21317 for checking signed/unsigned int16 <-> signed/unsigned int16 with
21318 depth 8, one for signed int16 <-> unsigned int16 and one for the new
21319 rounding from signed int32 to signed/unsigned int16.
21321 2007-03-27 11:31:17 +0000 Michael Smith <msmith@xiph.org>
21323 gst/audioconvert/gstaudioconvert.c: Fix typo in debug line introduced recently, as pointed out on irc.
21324 Original commit message from CVS:
21325 * gst/audioconvert/gstaudioconvert.c: (strip_width_64),
21326 (gst_audio_convert_transform_caps):
21327 Fix typo in debug line introduced recently, as pointed out on irc.
21329 2007-03-27 10:17:16 +0000 Tim-Philipp Müller <tim@centricular.net>
21331 Make sure we parse floating-point numbers in vorbis comments correctly with either '.' or ',' as separator, no matter...
21332 Original commit message from CVS:
21333 * gst-libs/gst/tag/gstvorbistag.c: (gst_vorbis_tag_add):
21334 * tests/check/libs/tag.c: (GST_START_TEST):
21335 Make sure we parse floating-point numbers in vorbis comments
21336 correctly with either '.' or ',' as separator, no matter what
21337 the current locale is. Add unit test for this too.
21339 2007-03-27 09:37:42 +0000 Thomas Vander Stichele <thomas@apestaart.org>
21341 * tests/check/pipelines/vorbisdec.c:
21343 Original commit message from CVS:
21346 2007-03-26 22:38:19 +0000 René Stadler <mail@renestadler.de>
21348 gst-libs/gst/tag/gstvorbistag.c: When writing out floating-point numbers to vorbis comment tags, always use the same ...
21349 Original commit message from CVS:
21350 Patch by: René Stadler <mail at renestadler de>
21351 * gst-libs/gst/tag/gstvorbistag.c: (gst_tag_to_vorbis_comments):
21352 When writing out floating-point numbers to vorbis comment tags, always
21353 use the same character as separator no matter what the current locale is
21355 * tests/check/libs/tag.c: (GST_START_TEST):
21356 Add unit tests for replaygain tags in vorbis comments (closes #423055).
21358 2007-03-26 20:56:35 +0000 Thomas Vander Stichele <thomas@apestaart.org>
21360 ext/vorbis/vorbisdec.c (vorbis_dec_push_forward, vorbis_handle_data_packet):
21361 Original commit message from CVS:
21362 * ext/vorbis/vorbisdec.c (vorbis_dec_push_forward,
21363 vorbis_handle_data_packet):
21364 Correctly set DURATION to generate a timestamp-continuous stream.
21365 One bug left at the end; see
21366 ihttp://bugzilla.gnome.org/show_bug.cgi?id=423086
21367 * tests/check/Makefile.am:
21368 * tests/check/pipelines/vorbisenc.c (GST_START_TEST):
21369 Add a test to check this. Without the above patch this test fails.
21371 2007-03-26 11:44:07 +0000 Jan Schmidt <thaytan@mad.scientist.com>
21373 gst-libs/gst/rtp/Makefile.am: The base audio payloader uses GstAdapter - we need GST_BASE_LIBS.
21374 Original commit message from CVS:
21375 * gst-libs/gst/rtp/Makefile.am:
21376 The base audio payloader uses GstAdapter - we need GST_BASE_LIBS.
21378 2007-03-23 15:43:24 +0000 Christian Schaller <uraeus@gnome.org>
21380 * gst-plugins-base.spec.in:
21382 Original commit message from CVS:
21385 2007-03-23 12:32:33 +0000 Michael Smith <msmith@xiph.org>
21387 gst/videorate/gstvideorate.c: If videorate changes caps, we can no longer use the old buffer (which may have a differ...
21388 Original commit message from CVS:
21389 * gst/videorate/gstvideorate.c: (gst_video_rate_setcaps),
21390 (gst_video_rate_reset), (gst_video_rate_chain):
21391 If videorate changes caps, we can no longer use the old buffer
21392 (which may have a different size, incompatible with our caps).
21393 So don't do that; just duplicate the new frame more times.
21395 2007-03-22 17:43:52 +0000 Jan Schmidt <thaytan@mad.scientist.com>
21397 gst/playback/gstplaybin.c: Remove playbin's override of the set_clock vmethod. It's irrelevant after Wim's commit on ...
21398 Original commit message from CVS:
21399 * gst/playback/gstplaybin.c: (gst_play_bin_class_init):
21400 Remove playbin's override of the set_clock vmethod. It's irrelevant
21401 after Wim's commit on the 19th.
21403 2007-03-22 14:37:08 +0000 Thomas Vander Stichele <thomas@apestaart.org>
21405 gst-libs/gst/app/Makefile.am: Use GST_ALL_LDFLAGS, which actually exists, but maybe David can confirm that was what h...
21406 Original commit message from CVS:
21407 * gst-libs/gst/app/Makefile.am:
21408 Use GST_ALL_LDFLAGS, which actually exists, but maybe David
21409 can confirm that was what he wanted.
21411 2007-03-22 09:26:02 +0000 Wim Taymans <wim.taymans@gmail.com>
21413 ext/gnomevfs/gstgnomevfssrc.*: Don't cache file sizes. Fixes #341078.
21414 Original commit message from CVS:
21415 * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_get_size),
21416 (gst_gnome_vfs_src_start), (gst_gnome_vfs_src_stop):
21417 * ext/gnomevfs/gstgnomevfssrc.h:
21418 Don't cache file sizes. Fixes #341078.
21420 2007-03-21 11:03:23 +0000 Tim-Philipp Müller <tim@centricular.net>
21422 gst/playback/gstplaybin.c: Use GST_PTR_FORMAT to log caps.
21423 Original commit message from CVS:
21424 * gst/playback/gstplaybin.c: (add_sink):
21425 Use GST_PTR_FORMAT to log caps.
21427 2007-03-21 10:23:11 +0000 Young-Ho Cha <ganadist@chollian.net>
21429 gst/subparse/samiparse.c: Special-case some more colour names that pango doesn't handle by default. Fixes #420578.
21430 Original commit message from CVS:
21431 Patch by: Young-Ho Cha <ganadist at chollian net>
21432 * gst/subparse/samiparse.c: (handle_start_font):
21433 Special-case some more colour names that pango doesn't handle by
21434 default. Fixes #420578.
21436 2007-03-20 11:49:55 +0000 Michael Smith <msmith@xiph.org>
21438 ext/vorbis/vorbisenc.c: If we get a zero-sized input buffer, don't pass it to libvorbis, as that marks EOS internally...
21439 Original commit message from CVS:
21440 * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_chain):
21441 If we get a zero-sized input buffer, don't pass it to libvorbis, as
21442 that marks EOS internally. After that, libvorbis will buffer all
21443 input data, and encode none of it, eventually leading to memory
21446 2007-03-19 10:52:50 +0000 Wim Taymans <wim.taymans@gmail.com>
21448 gst/playback/gstdecodebin.c: Don't post STATE_DIRTY anymore.
21449 Original commit message from CVS:
21450 * gst/playback/gstdecodebin.c: (remove_fakesink):
21451 Don't post STATE_DIRTY anymore.
21452 * gst/playback/gstplaybin.c: (add_sink), (gst_play_bin_send_event),
21453 (gst_play_bin_change_state):
21454 Remove stream_time reset in seek handling, core does that now.
21455 Disable clocking for live pipelines by forcing a NULL clock to the
21456 complete pipeline, core is too smart now for our previous hack.
21457 We can always autoplug in PAUSED now.
21459 2007-03-18 03:14:01 +0000 David Schleef <ds@schleef.org>
21461 REQUIREMENTS: Update this file, change the formatting to make it more consistent, plus more machine readable.
21462 Original commit message from CVS:
21463 * REQUIREMENTS: Update this file, change the formatting to make
21464 it more consistent, plus more machine readable.
21466 2007-03-16 17:29:09 +0000 Michael Smith <msmith@xiph.org>
21468 gst/audioconvert/gstaudioconvert.c: Previous fix was too simplistic, and broke the tests. Use a better approach; only...
21469 Original commit message from CVS:
21470 * gst/audioconvert/gstaudioconvert.c: (make_lossless_changes),
21471 (strip_width_64), (append_with_other_format):
21472 Previous fix was too simplistic, and broke the tests. Use a better
21473 approach; only strip 64 from widths for integer audio.
21475 2007-03-16 16:42:23 +0000 Michael Smith <msmith@xiph.org>
21477 gst/audioconvert/gstaudioconvert.c: We don't support 64 bit integer audio, so don't try to claim we can.
21478 Original commit message from CVS:
21479 * gst/audioconvert/gstaudioconvert.c: (make_lossless_changes),
21480 (gst_audio_convert_transform_caps):
21481 We don't support 64 bit integer audio, so don't try to claim we can.
21482 Stops us producing caps don't match our template caps.
21485 2007-03-15 10:52:21 +0000 Michael Smith <msmith@xiph.org>
21487 gst/audioresample/gstaudioresample.c: Don't trigger discontinuities for very small imperfections; a filter flush will...
21488 Original commit message from CVS:
21489 * gst/audioresample/gstaudioresample.c:
21490 (audioresample_check_discont), (audioresample_transform):
21491 Don't trigger discontinuities for very small imperfections; a filter
21492 flush will sound bad, and many plugins have rounding errors leading
21495 2007-03-14 21:11:18 +0000 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
21497 gst-libs/gst/rtp/gstbasertpaudiopayload.*: olivier.crete@collabora.co.uk.
21498 Original commit message from CVS:
21499 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
21500 * gst-libs/gst/rtp/gstbasertpaudiopayload.h:
21501 Add min-ptime property to RTP base audio payloader. Patch by
21502 olivier.crete@collabora.co.uk.
21504 Indentation/whitespace/documentation fixes.
21506 2007-03-14 17:16:30 +0000 Julien Moutte <julien@moutte.net>
21508 gst/audioresample/gstaudioresample.c: Handle discontinuous streams.
21509 Original commit message from CVS:
21510 2007-03-14 Julien MOUTTE <julien@moutte.net>
21511 * gst/audioresample/gstaudioresample.c: (gst_audioresample_init),
21512 (audioresample_transform_size), (audioresample_do_output),
21513 (audioresample_transform), (audioresample_pushthrough): Handle
21514 discontinuous streams.
21515 * gst/audioresample/gstaudioresample.h:
21516 * tests/check/elements/audioresample.c:
21517 (test_discont_stream_instance), (GST_START_TEST),
21518 (audioresample_suite): Add a test for discontinuous streams.
21519 * win32/common/config.h: Updated.
21521 2007-03-14 15:16:23 +0000 Thomas Vander Stichele <thomas@apestaart.org>
21523 po/: Update translations from translation project.
21524 Original commit message from CVS:
21538 Update translations from translation project.
21540 2007-03-14 15:05:32 +0000 Thomas Vander Stichele <thomas@apestaart.org>
21542 * gst/gdp/gstgdpdepay.c:
21544 Original commit message from CVS:
21547 2007-03-14 14:48:12 +0000 Thomas Vander Stichele <thomas@apestaart.org>
21549 gst/audioresample/: Since I really am not interested in a debug line for each sample being processed, move the librar...
21550 Original commit message from CVS:
21551 * gst/audioresample/debug.h:
21552 * gst/audioresample/resample.c: (resample_init):
21553 Since I really am not interested in a debug line for each sample
21554 being processed, move the library's debugging to its own category,
21557 2007-03-14 14:09:21 +0000 Thomas Vander Stichele <thomas@apestaart.org>
21559 * gst/audioresample/gstaudioresample.c:
21560 add debugging and reformat docs
21561 Original commit message from CVS:
21562 add debugging and reformat docs
21564 2007-03-12 23:29:07 +0000 Michael Smith <msmith@xiph.org>
21566 ext/theora/theoradec.c: Since the plugin doesn't support anything other than 4:2:0 right now, post an error and fail ...
21567 Original commit message from CVS:
21568 * ext/theora/theoradec.c: (theora_handle_type_packet):
21569 Since the plugin doesn't support anything other than 4:2:0 right
21570 now, post an error and fail if we get something else. Won't matter
21571 until libtheora supports the other pixel formats, but hopefully
21574 2007-03-12 15:50:35 +0000 Alex Lancaster <alexlan@fedoraproject.org>
21577 I'm too lazy to comment this
21578 Original commit message from CVS:
21579 Mention Patch by: Alex Lancaster in a recent commit.
21581 2007-03-12 11:47:42 +0000 Jan Schmidt <thaytan@mad.scientist.com>
21583 examples/app/.cvsignore: The buildbot demands .cvsignore files, and I comply.
21584 Original commit message from CVS:
21585 * examples/app/.cvsignore:
21586 The buildbot demands .cvsignore files, and I comply.
21588 2007-03-11 00:48:26 +0000 David Schleef <ds@schleef.org>
21590 Add appsrc/appsink example.
21591 Original commit message from CVS:
21593 * examples/Makefile.am:
21594 * examples/app/Makefile.am:
21595 * examples/app/appsrc_ex.c:
21596 Add appsrc/appsink example.
21597 * gst-libs/gst/app/Makefile.am:
21598 * gst-libs/gst/app/gstapp.c:
21599 * gst-libs/gst/app/gstappsink.c:
21600 * gst-libs/gst/app/gstappsink.h:
21601 * gst/app/gstapp.c:
21604 2007-03-10 15:59:33 +0000 Sébastien Moutte <sebastien@moutte.net>
21606 gst-libs/gst/audio/gstbaseaudiosink.c: Use gst_guint64_to_gdouble for conversion.
21607 Original commit message from CVS:
21608 * gst-libs/gst/audio/gstbaseaudiosink.c:(gst_base_audio_sink_render):
21609 Use gst_guint64_to_gdouble for conversion.
21611 Add new files to the win32 MANIFEST.
21612 * win32/common/libgstaudio.def:
21613 * win32/common/libgstpbutils.def:
21614 Add new exported functions.
21615 * win32/vs6/gst_plugins_base.dsw:
21616 * win32/vs6/libgstdecodebin.dsp:
21617 * win32/vs6/libgstplaybin.dsp:
21618 Change the link to libgstpbutils.lib.
21619 * win32/vs6/libgstdecodebin2.dsp:
21620 Add a new project for decodebin2.
21621 * win32/vs6/libgstpbutils.dsp:
21622 Add a new project for pbutils.
21624 2007-03-10 12:18:58 +0000 Tim-Philipp Müller <tim@centricular.net>
21626 gst-libs/gst/tag/gstvorbistag.c: Also accept partial dates with only year and month, like 1999-12-00 (fixes #410396 e...
21627 Original commit message from CVS:
21628 * gst-libs/gst/tag/gstvorbistag.c: (gst_vorbis_tag_add):
21629 Also accept partial dates with only year and month,
21630 like 1999-12-00 (fixes #410396 even more).
21631 * tests/check/libs/tag.c: (GST_START_TEST):
21632 Add unit test for the above.
21634 2007-03-10 11:21:08 +0000 Tim-Philipp Müller <tim@centricular.net>
21636 tests/check/elements/subparse.c: Add unit test for MPL2 subtitle format (#413799).
21637 Original commit message from CVS:
21638 * tests/check/elements/subparse.c: (GST_START_TEST),
21640 Add unit test for MPL2 subtitle format (#413799).
21642 2007-03-10 11:17:52 +0000 Kamil Pawlowski <kamilpe@gmail.com>
21644 gst/subparse/: Add support for MPL2 subtitle format (#413799).
21645 Original commit message from CVS:
21646 Patch by: Kamil Pawlowski <kamilpe gmail com>
21647 * gst/subparse/Makefile.am:
21648 * gst/subparse/gstsubparse.c:
21649 (gst_sub_parse_data_format_autodetect),
21650 (gst_sub_parse_format_autodetect), (gst_sub_parse_sink_event),
21651 (gst_subparse_type_find):
21652 * gst/subparse/gstsubparse.h:
21653 * gst/subparse/mpl2parse.c: (mpl2_parse_line), (parse_mpl2):
21654 * gst/subparse/mpl2parse.h:
21655 Add support for MPL2 subtitle format (#413799).
21657 2007-03-09 17:33:17 +0000 Tim-Philipp Müller <tim@centricular.net>
21659 configure.ac: We require core CVS for the new buffer metadata copy functions.
21660 Original commit message from CVS:
21662 We require core CVS for the new buffer metadata copy functions.
21664 2007-03-09 16:51:13 +0000 Wim Taymans <wim.taymans@gmail.com>
21666 gst-libs/gst/tag/gstid3tag.c: Add read support for GST_TAG_MUSICBRAINZ_SORTNAME (TSOP) tag.
21667 Original commit message from CVS:
21668 * gst-libs/gst/tag/gstid3tag.c:
21669 Add read support for GST_TAG_MUSICBRAINZ_SORTNAME (TSOP) tag.
21672 2007-03-09 16:46:35 +0000 Wim Taymans <wim.taymans@gmail.com>
21674 ext/libvisual/visual.c: Improve adapter usage and comments.
21675 Original commit message from CVS:
21676 * ext/libvisual/visual.c: (gst_visual_sink_setcaps),
21677 (gst_vis_src_negotiate), (get_buffer), (gst_visual_chain):
21678 Improve adapter usage and comments.
21680 2007-03-09 16:38:06 +0000 Wim Taymans <wim.taymans@gmail.com>
21682 Use new metadata copy function.
21683 Original commit message from CVS:
21684 * ext/pango/gsttextrender.c: (gst_text_render_chain):
21685 * ext/vorbis/vorbistag.c: (gst_vorbis_tag_parse_packet):
21686 * gst-libs/gst/netbuffer/gstnetbuffer.c: (gst_netbuffer_copy):
21687 Use new metadata copy function.
21688 * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
21689 (gst_ffmpegcsp_transform):
21690 * gst/videoscale/gstvideoscale.c: (gst_video_scale_transform):
21691 Basetransform copied the metadata for us.
21693 2007-03-09 16:28:04 +0000 Tim-Philipp Müller <tim@centricular.net>
21695 ext/pango/gsttextoverlay.c: Some more logging. Only accept newsegment events in TIME format and send a WARNING messag...
21696 Original commit message from CVS:
21697 * ext/pango/gsttextoverlay.c: (gst_text_overlay_text_event),
21698 (gst_text_overlay_video_event):
21699 Some more logging. Only accept newsegment events in TIME format and
21700 send a WARNING message if they are not in TIME format.
21701 * gst/subparse/gstsubparse.c: (gst_sub_parse_dispose),
21702 (gst_sub_parse_init), (gst_sub_parse_src_event), (handle_buffer),
21703 (gst_sub_parse_chain), (gst_sub_parse_sink_event):
21704 * gst/subparse/gstsubparse.h:
21705 No need to allocate GstSegment structure dynamically, just put it
21706 into the instance structure; ignore newsegment events in BYTE
21707 format and in particular don't let it overwrite our saved TIME
21708 segment from the last seek.
21710 2007-03-09 13:05:04 +0000 Michael Smith <msmith@xiph.org>
21712 gst/typefind/gsttypefindfunctions.c: Replace AC3 typefinder with one that isn't terrible, and actually works usefully.
21713 Original commit message from CVS:
21714 * gst/typefind/gsttypefindfunctions.c: (ac3_type_find):
21715 Replace AC3 typefinder with one that isn't terrible, and actually
21718 2007-03-09 12:22:53 +0000 Thomas Vander Stichele <thomas@apestaart.org>
21720 gst/audioconvert/gstaudioconvert.c: fix error category and translatable string
21721 Original commit message from CVS:
21722 * gst/audioconvert/gstaudioconvert.c:
21723 (gst_audio_convert_transform):
21724 fix error category and translatable string
21726 2007-03-09 11:23:32 +0000 Tim-Philipp Müller <tim@centricular.net>
21728 pkgconfig/: Fix up utils => pbutils here too.
21729 Original commit message from CVS:
21730 * pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
21731 * pkgconfig/gstreamer-plugins-base.pc.in:
21732 Fix up utils => pbutils here too.
21734 2007-03-09 10:49:53 +0000 Tim-Philipp Müller <tim@centricular.net>
21736 gst/subparse/gstsubparse.c: Break out of loop in chain function as soon as possible if we get a non-OK flow return.
21737 Original commit message from CVS:
21738 * gst/subparse/gstsubparse.c: (handle_buffer):
21739 Break out of loop in chain function as soon as possible if we get
21740 a non-OK flow return.
21742 2007-03-08 18:26:07 +0000 Jan Schmidt <thaytan@mad.scientist.com>
21744 tests/check/elements/alsa.c: Unref the mixer if the state change fails too (if the alsa devices are inaccessible, for...
21745 Original commit message from CVS:
21746 * tests/check/elements/alsa.c: (GST_START_TEST):
21747 Unref the mixer if the state change fails too (if the
21748 alsa devices are inaccessible, for example)
21750 2007-03-08 17:49:46 +0000 Jan Schmidt <thaytan@mad.scientist.com>
21752 tests/check/Makefile.am: Don't test libvisual elements in the states check, because libvisual seems to leak internally.
21753 Original commit message from CVS:
21754 * tests/check/Makefile.am:
21755 Don't test libvisual elements in the states check, because libvisual
21756 seems to leak internally.
21757 Re-enable the alsa and states tests now that there's new suppressions
21759 * tests/check/elements/alsa.c: (GST_START_TEST):
21760 Don't leak the alsamixer we instantiated.
21762 2007-03-08 15:22:53 +0000 Jan Schmidt <thaytan@mad.scientist.com>
21764 sys/: Move some cleanup stuff from the state change handler into a _reset() function that can be called from _finaliz...
21765 Original commit message from CVS:
21766 * sys/ximage/ximagesink.c: (gst_ximagesink_xcontext_clear),
21767 (gst_ximagesink_change_state), (gst_ximagesink_reset),
21768 (gst_ximagesink_finalize):
21769 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_change_state),
21770 (gst_xvimagesink_reset), (gst_xvimagesink_finalize):
21771 Move some cleanup stuff from the state change handler into a _reset()
21772 function that can be called from _finalize(). This ensures that things
21773 get freed even if (for some reason) the NULL->READY state transition
21774 fails in the parent class.
21775 Even if a parent state change fails, process our downward state change
21776 logic instead of bailing out early.
21777 Free the correct xcontext pointer in ximagesink's xcontext_clear.
21779 2007-03-08 12:53:51 +0000 Jan Schmidt <thaytan@mad.scientist.com>
21781 ext/alsa/gstalsasink.c: Extra log line.
21782 Original commit message from CVS:
21783 * ext/alsa/gstalsasink.c: (gst_alsasink_open):
21785 * ext/pango/gstclockoverlay.c: (gst_clock_overlay_init):
21786 * ext/pango/gsttimeoverlay.c: (gst_time_overlay_init):
21787 Use pango_font_description_set_family_static instead of
21788 pango_font_description_set_family to save a string copy (it was
21789 leaking due to the strdup anyway)
21790 * gst/tcp/gsttcpclientsink.c: (gst_tcp_client_sink_finalize):
21791 * gst/tcp/gsttcpclientsrc.c: (gst_tcp_client_src_finalize):
21792 * gst/tcp/gsttcpserversink.c: (gst_tcp_server_sink_finalize):
21793 * gst/tcp/gsttcpserversrc.c: (gst_tcp_server_src_finalize):
21794 Chain up in finalize.
21796 2007-03-07 18:50:10 +0000 Tim-Philipp Müller <tim@centricular.net>
21798 gst-libs/gst/interfaces/mixertrack.c: API: add "untranslated-label" property which should be set by implementations a...
21799 Original commit message from CVS:
21800 * gst-libs/gst/interfaces/mixertrack.c:
21801 (gst_mixer_track_class_init), (gst_mixer_track_get_property),
21802 (gst_mixer_track_set_property):
21803 API: add "untranslated-label" property which should be set by
21804 implementations at construct time (#414645).
21805 * ext/alsa/gstalsamixeroptions.c: (gst_alsa_mixer_options_new):
21806 * ext/alsa/gstalsamixertrack.c: (gst_alsa_mixer_track_new):
21807 Set "untranslated-label" when constructing mixer track objects.
21808 * tests/check/elements/alsa.c: (GST_START_TEST), (alsa_suite):
21809 Unit test to check the above.
21811 2007-03-07 17:15:57 +0000 Wim Taymans <wim.taymans@gmail.com>
21813 ext/ogg/gstoggdemux.c: Fix confusing debug message.
21814 Original commit message from CVS:
21815 * ext/ogg/gstoggdemux.c: (gst_ogg_demux_read_chain):
21816 Fix confusing debug message.
21818 2007-03-07 17:12:54 +0000 Jan Schmidt <thaytan@mad.scientist.com>
21820 gst-plugins-base.doap: update doap file with new version
21821 Original commit message from CVS:
21822 * gst-plugins-base.doap:
21823 update doap file with new version
21825 2007-03-07 17:05:21 +0000 Thomas Vander Stichele <thomas@apestaart.org>
21827 * gst/tcp/gstmultifdsink.c:
21829 Original commit message from CVS:
21832 2007-03-07 16:56:01 +0000 Jan Schmidt <thaytan@mad.scientist.com>
21834 configure.ac: Back to CVS
21835 Original commit message from CVS:
21839 === release 0.10.12 ===
21841 2007-03-07 16:46:51 +0000 Jan Schmidt <thaytan@mad.scientist.com>
21847 * docs/plugins/gst-plugins-base-plugins.args:
21848 * docs/plugins/inspect/plugin-adder.xml:
21849 * docs/plugins/inspect/plugin-alsa.xml:
21850 * docs/plugins/inspect/plugin-audioconvert.xml:
21851 * docs/plugins/inspect/plugin-audiorate.xml:
21852 * docs/plugins/inspect/plugin-audioresample.xml:
21853 * docs/plugins/inspect/plugin-audiotestsrc.xml:
21854 * docs/plugins/inspect/plugin-cdparanoia.xml:
21855 * docs/plugins/inspect/plugin-decodebin.xml:
21856 * docs/plugins/inspect/plugin-decodebin2.xml:
21857 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
21858 * docs/plugins/inspect/plugin-gdp.xml:
21859 * docs/plugins/inspect/plugin-gnomevfs.xml:
21860 * docs/plugins/inspect/plugin-libvisual.xml:
21861 * docs/plugins/inspect/plugin-ogg.xml:
21862 * docs/plugins/inspect/plugin-pango.xml:
21863 * docs/plugins/inspect/plugin-playbin.xml:
21864 * docs/plugins/inspect/plugin-subparse.xml:
21865 * docs/plugins/inspect/plugin-tcp.xml:
21866 * docs/plugins/inspect/plugin-theora.xml:
21867 * docs/plugins/inspect/plugin-typefindfunctions.xml:
21868 * docs/plugins/inspect/plugin-video4linux.xml:
21869 * docs/plugins/inspect/plugin-videorate.xml:
21870 * docs/plugins/inspect/plugin-videoscale.xml:
21871 * docs/plugins/inspect/plugin-videotestsrc.xml:
21872 * docs/plugins/inspect/plugin-volume.xml:
21873 * docs/plugins/inspect/plugin-vorbis.xml:
21874 * docs/plugins/inspect/plugin-ximagesink.xml:
21875 * docs/plugins/inspect/plugin-xvimagesink.xml:
21876 * win32/common/config.h:
21878 Original commit message from CVS:
21881 2007-03-07 15:35:26 +0000 Jan Schmidt <thaytan@mad.scientist.com>
21900 Original commit message from CVS:
21903 2007-03-06 12:31:01 +0000 Jan Schmidt <thaytan@mad.scientist.com>
21905 configure.ac: Bump version to 0.10.11.4 pre-release
21906 Original commit message from CVS:
21908 Bump version to 0.10.11.4 pre-release
21910 2007-03-06 12:10:08 +0000 Wim Taymans <wim.taymans@gmail.com>
21912 gst-libs/gst/audio/gstbaseaudiosink.c: Fix regression that made GStreamer skip the first samples of audio.
21913 Original commit message from CVS:
21914 * gst-libs/gst/audio/gstbaseaudiosink.c:
21915 (gst_base_audio_sink_async_play):
21916 Fix regression that made GStreamer skip the first samples of audio.
21919 2007-03-05 11:21:13 +0000 Jan Schmidt <thaytan@mad.scientist.com>
21921 configure.ac: Bump version to 0.10.11.3 pre-release
21922 Original commit message from CVS:
21924 Bump version to 0.10.11.3 pre-release
21926 2007-03-05 09:35:29 +0000 Sebastian Dröge <slomo@circular-chaos.org>
21928 po/POTFILES.in: Update paths for the rename from utils to pbutils to fix the build.
21929 Original commit message from CVS:
21931 Update paths for the rename from utils to pbutils to fix the build.
21933 2007-03-05 09:27:55 +0000 Tim-Philipp Müller <tim@centricular.net>
21935 gst-libs/gst/pbutils/Makefile.am: Change directory to install headers in from gst/utils to gst/pbutils as well.
21936 Original commit message from CVS:
21937 * gst-libs/gst/pbutils/Makefile.am:
21938 Change directory to install headers in from gst/utils to gst/pbutils
21941 2007-03-04 23:41:51 +0000 Thomas Vander Stichele <thomas@apestaart.org>
21943 * tests/check/libs/.gitignore:
21945 Original commit message from CVS:
21948 2007-03-04 23:41:04 +0000 Thomas Vander Stichele <thomas@apestaart.org>
21950 * win32/common/config.h:
21951 * win32/common/libgstutils.def:
21953 Original commit message from CVS:
21956 2007-03-04 23:39:51 +0000 Thomas Vander Stichele <thomas@apestaart.org>
21958 rename utils to pbutils
21959 Original commit message from CVS:
21961 * docs/libs/gst-plugins-base-libs-docs.sgml:
21962 * docs/libs/gst-plugins-base-libs-sections.txt:
21963 * gst-libs/gst/Makefile.am:
21964 * gst-libs/gst/interfaces/mixer.c:
21965 * gst-libs/gst/pbutils/Makefile.am:
21966 * gst-libs/gst/pbutils/descriptions.c:
21967 (gst_pb_utils_get_source_description),
21968 (gst_pb_utils_get_sink_description),
21969 (gst_pb_utils_get_decoder_description),
21970 (gst_pb_utils_get_encoder_description),
21971 (gst_pb_utils_get_element_description),
21972 (gst_pb_utils_add_codec_description_to_tag_list),
21973 (gst_pb_utils_get_codec_description), (gst_pb_utils_list_all):
21974 * gst-libs/gst/pbutils/descriptions.h:
21975 * gst-libs/gst/pbutils/install-plugins.c:
21976 * gst-libs/gst/pbutils/install-plugins.h:
21977 * gst-libs/gst/pbutils/missing-plugins.c:
21978 (gst_missing_uri_source_message_new),
21979 (gst_missing_uri_sink_message_new),
21980 (gst_missing_element_message_new),
21981 (gst_missing_decoder_message_new),
21982 (gst_missing_encoder_message_new),
21983 (gst_missing_plugin_message_get_description):
21984 * gst-libs/gst/pbutils/missing-plugins.h:
21985 * gst-libs/gst/pbutils/pbutils.c: (gst_pb_utils_init):
21986 * gst-libs/gst/pbutils/pbutils.h:
21987 * gst-libs/gst/utils/Makefile.am:
21988 * gst-libs/gst/utils/base-utils.c:
21989 * gst-libs/gst/utils/base-utils.h:
21990 * gst-libs/gst/utils/descriptions.c:
21991 * gst-libs/gst/utils/descriptions.h:
21992 * gst-libs/gst/utils/install-plugins.c:
21993 * gst-libs/gst/utils/install-plugins.h:
21994 * gst-libs/gst/utils/missing-plugins.c:
21995 * gst-libs/gst/utils/missing-plugins.h:
21996 * gst-plugins-base.spec.in:
21997 * gst/playback/Makefile.am:
21998 * gst/playback/gstdecodebin.c:
21999 * gst/playback/gstdecodebin2.c:
22000 * gst/playback/gstplaybasebin.c: (setup_subtitle),
22001 (gen_source_element):
22002 * gst/playback/gstplaybin.c: (plugin_init):
22003 * tests/check/Makefile.am:
22004 * tests/check/libs/pbutils.c: (GST_START_TEST),
22005 (test_pb_utils_install_plugins_do_callout), (libgstpbutils_suite):
22006 * tests/check/libs/utils.c:
22007 rename utils to pbutils
22009 2007-03-03 10:23:03 +0000 David Schleef <ds@schleef.org>
22011 gst-libs/gst/app/Makefile.am: Install the headers.
22012 Original commit message from CVS:
22013 * gst-libs/gst/app/Makefile.am:
22014 Install the headers.
22016 2007-03-03 10:10:30 +0000 David Schleef <ds@schleef.org>
22018 gst-libs/gst/app/: Add GstAppBuffer that includes a callback and closure for proper handling of data chunks.
22019 Original commit message from CVS:
22020 * gst-libs/gst/app/Makefile.am:
22021 * gst-libs/gst/app/gstappbuffer.c:
22022 * gst-libs/gst/app/gstappbuffer.h:
22023 * gst-libs/gst/app/gstappsrc.c:
22024 Add GstAppBuffer that includes a callback and closure for
22025 proper handling of data chunks.
22027 2007-03-03 09:06:06 +0000 David Schleef <ds@schleef.org>
22029 gst-libs/gst/app/gstappsrc.*: Hacking to address issues in 413418.
22030 Original commit message from CVS:
22031 * gst-libs/gst/app/gstappsrc.c:
22032 * gst-libs/gst/app/gstappsrc.h:
22033 Hacking to address issues in 413418.
22035 2007-03-03 08:16:57 +0000 David Schleef <ds@schleef.org>
22037 Move the app library to gst-libs/gst/app (duh!)
22038 Original commit message from CVS:
22042 * gst-libs/gst/Makefile.am:
22043 * gst-libs/gst/app/Makefile.am:
22044 * gst-libs/gst/app/gstapp.c:
22045 * gst-libs/gst/app/gstappsrc.c:
22046 * gst-libs/gst/app/gstappsrc.h:
22047 * gst/app/Makefile.am:
22048 * gst/app/gstapp.c:
22049 * gst/app/gstappsrc.c:
22050 * gst/app/gstappsrc.h:
22051 Move the app library to gst-libs/gst/app (duh!)
22053 2007-03-02 12:59:15 +0000 Jan Schmidt <thaytan@mad.scientist.com>
22055 Add documentation for decodebin2 that indicates that the API is still unstable.
22056 Original commit message from CVS:
22057 * docs/plugins/Makefile.am:
22058 * docs/plugins/gst-plugins-base-plugins-docs.sgml:
22059 * docs/plugins/gst-plugins-base-plugins-sections.txt:
22060 * docs/plugins/inspect/plugin-decodebin2.xml:
22061 * gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init):
22062 Add documentation for decodebin2 that indicates that the API
22065 2007-03-01 18:50:00 +0000 Jan Schmidt <thaytan@mad.scientist.com>
22067 configure.ac: Update to 0.10.11.2 (0.10.12 pre-release)
22068 Original commit message from CVS:
22070 Update to 0.10.11.2 (0.10.12 pre-release)
22072 2007-03-01 17:29:55 +0000 Wim Taymans <wim.taymans@gmail.com>
22074 gst-libs/gst/audio/gstbaseaudiosink.c: base time is irrelevant here.
22075 Original commit message from CVS:
22076 * gst-libs/gst/audio/gstbaseaudiosink.c:
22077 (gst_base_audio_sink_async_play):
22078 base time is irrelevant here.
22080 2007-03-01 17:01:43 +0000 Wim Taymans <wim.taymans@gmail.com>
22082 gst-libs/gst/audio/: Improve debugging.
22083 Original commit message from CVS:
22084 * gst-libs/gst/audio/gstaudiosink.c: (audioringbuffer_thread_func):
22085 * gst-libs/gst/audio/gstaudiosrc.c: (audioringbuffer_thread_func):
22087 * gst-libs/gst/audio/gstbaseaudiosink.c:
22088 (gst_base_audio_sink_query), (gst_base_audio_sink_event),
22089 (gst_base_audio_sink_render), (gst_base_audio_sink_async_play):
22090 Improve latency and clock slaving calculations.
22091 Improve slave clock calibration.
22092 * gst-libs/gst/audio/gstringbuffer.c:
22093 (gst_ring_buffer_commit_full):
22094 When we are asked to render N sample to 0 bytes, return N.
22096 2007-03-01 16:48:45 +0000 Wim Taymans <wim.taymans@gmail.com>
22098 ext/alsa/gstalsasink.*: Remove unused dispose function.
22099 Original commit message from CVS:
22100 * ext/alsa/gstalsasink.c: (gst_alsasink_class_init),
22101 (gst_alsasink_write), (gst_alsasink_reset):
22102 * ext/alsa/gstalsasink.h:
22103 Remove unused dispose function.
22104 Rename lock to not interfere with alsasrc lock.
22105 * ext/alsa/gstalsasrc.c: (gst_alsasrc_finalize),
22106 (gst_alsasrc_class_init), (gst_alsasrc_init), (set_swparams),
22107 (gst_alsasrc_read), (gst_alsasrc_reset):
22108 * ext/alsa/gstalsasrc.h:
22109 Implement finalize function.
22110 Use lock to protect alsa access.
22112 Fine tune sw params.
22114 2007-03-01 10:20:59 +0000 Thomas Vander Stichele <thomas@apestaart.org>
22119 Original commit message from CVS:
22122 2007-02-28 19:27:28 +0000 Thomas Vander Stichele <thomas@apestaart.org>
22124 configure.ac: Convert to new AG_GST style.
22125 Original commit message from CVS:
22127 Convert to new AG_GST style.
22129 2007-02-28 15:17:20 +0000 Ed Catmur <ed@catmur.co.uk>
22131 gst/playback/gstplaybin.c: Fix race condition when rapidly switching visualisations in playbin.
22132 Original commit message from CVS:
22133 Patch by: Ed Catmur <ed at catmur dot co dot uk>
22134 * gst/playback/gstplaybin.c: (gst_play_bin_vis_unblocked),
22135 (gst_play_bin_vis_blocked), (gst_play_bin_set_property):
22136 Fix race condition when rapidly switching visualisations in playbin.
22139 2007-02-28 15:11:59 +0000 Jan Schmidt <thaytan@mad.scientist.com>
22141 tests/check/Makefile.am: Include local stuff before system installed things in LDFLAGS and
22142 Original commit message from CVS:
22143 * tests/check/Makefile.am:
22144 Include local stuff before system installed things in LDFLAGS and
22147 2007-02-28 15:10:06 +0000 Wim Taymans <wim.taymans@gmail.com>
22149 ext/ogg/gstoggdemux.c: Improve debugging.
22150 Original commit message from CVS:
22151 * ext/ogg/gstoggdemux.c: (gst_ogg_demux_sink_activate):
22154 2007-02-28 15:05:03 +0000 Wim Taymans <wim.taymans@gmail.com>
22156 sys/v4l/: Fix duration and timestamping, taking latency into account.
22157 Original commit message from CVS:
22158 * sys/v4l/gstv4lsrc.c: (gst_v4lsrc_class_init), (gst_v4lsrc_init),
22159 (gst_v4lsrc_fixate), (gst_v4lsrc_query):
22160 * sys/v4l/v4lsrc_calls.c: (gst_v4lsrc_buffer_new):
22161 Fix duration and timestamping, taking latency into account.
22162 Implement latency query.
22164 2007-02-28 15:02:25 +0000 Wim Taymans <wim.taymans@gmail.com>
22166 gst-libs/gst/audio/gstaudioclock.c: Fix clock name.
22167 Original commit message from CVS:
22168 * gst-libs/gst/audio/gstaudioclock.c: (gst_audio_clock_init),
22169 (gst_audio_clock_new):
22171 * gst-libs/gst/audio/gstbaseaudiosink.c:
22172 (gst_base_audio_sink_init), (gst_base_audio_sink_query):
22173 * gst-libs/gst/audio/gstbaseaudiosrc.c: (gst_base_audio_src_init),
22174 (gst_base_audio_src_query), (gst_base_audio_src_get_offset),
22175 (gst_base_audio_src_create):
22176 Improve latency query code.
22177 Use proper clock names.
22179 2007-02-28 12:57:46 +0000 Thomas Vander Stichele <thomas@apestaart.org>
22181 * tests/check/generic/states.c:
22183 Original commit message from CVS:
22186 2007-02-28 12:44:53 +0000 Thomas Vander Stichele <thomas@apestaart.org>
22188 tests/check/generic/states.c: Copy the states.c test from core again
22189 Original commit message from CVS:
22190 * tests/check/generic/states.c: (GST_START_TEST):
22191 Copy the states.c test from core again
22192 * tests/check/Makefile.am:
22193 ignore cdio and cdparanoiasrc
22195 2007-02-28 12:08:27 +0000 Stefan Kost <ensonic@users.sourceforge.net>
22197 gst/audioconvert/audioconvert.c: Also make valgrind happy and avoid copying data in some cases.
22198 Original commit message from CVS:
22199 * gst/audioconvert/audioconvert.c: (float), (double), (float_hq),
22200 (double_hq), (audio_convert_get_func_index), (check_default),
22201 (audio_convert_prepare_context), (audio_convert_convert):
22202 Also make valgrind happy and avoid copying data in some cases.
22204 2007-02-28 11:58:16 +0000 Thomas Vander Stichele <thomas@apestaart.org>
22206 * tests/check/generic/states.c:
22208 Original commit message from CVS:
22211 2007-02-28 11:47:45 +0000 Stefan Kost <ensonic@users.sourceforge.net>
22213 Don't run inplace if that overwrites source data as we go. Add more tests. Fixes #339837 even more.
22214 Original commit message from CVS:
22215 * gst/audioconvert/audioconvert.c: (float), (double), (float_hq),
22216 (double_hq), (audio_convert_get_func_index),
22217 (audio_convert_prepare_context), (audio_convert_convert):
22218 * gst/audioconvert/gstaudioconvert.c:
22219 (gst_audio_convert_class_init), (gst_audio_convert_get_unit_size),
22220 (gst_audio_convert_transform_caps):
22221 * tests/check/elements/audioconvert.c: (GST_START_TEST),
22222 (audioconvert_suite):
22223 Don't run inplace if that overwrites source data as we go. Add more
22224 tests. Fixes #339837 even more.
22226 2007-02-27 18:45:37 +0000 Julien Moutte <julien@moutte.net>
22228 tests/examples/seek/seek.c: Fix various seeking bugs (Slider was not updating when doing a non flushing seek, Reverse...
22229 Original commit message from CVS:
22230 2007-02-27 Julien MOUTTE <julien@moutte.net>
22231 * tests/examples/seek/seek.c: (do_seek), (set_update_scale),
22232 (msg_segment_done): Fix various seeking bugs (Slider was not
22233 updating when doing a non flushing seek, Reverse playback
22234 on segment seek was wrong).
22236 2007-02-26 21:01:03 +0000 David Schleef <ds@schleef.org>
22238 Add a new plugin/library to make it easy for apps to shove data into a pipeline.
22239 Original commit message from CVS:
22241 * gst/app/Makefile.am:
22242 * gst/app/gstapp.c:
22243 * gst/app/gstappsrc.c:
22244 * gst/app/gstappsrc.h:
22245 Add a new plugin/library to make it easy for apps to shove
22246 data into a pipeline.
22248 2007-02-26 11:48:49 +0000 Wim Taymans <wim.taymans@gmail.com>
22250 tests/examples/seek/seek.c: When we stop scrubbing, don't leave the pipeline PLAYING when we requested a PAUSED state.
22251 Original commit message from CVS:
22252 * tests/examples/seek/seek.c: (stop_seek):
22253 When we stop scrubbing, don't leave the pipeline PLAYING when we
22254 requested a PAUSED state.
22256 2007-02-25 23:51:03 +0000 René Stadler <mail@renestadler.de>
22258 gst-libs/gst/tag/gstvorbistag.c: Parse date strings in vorbis comments that have an invalid (zero) month or day (#410...
22259 Original commit message from CVS:
22260 Patch by: René Stadler <mail at renestadler de>
22261 * gst-libs/gst/tag/gstvorbistag.c: (gst_vorbis_tag_add):
22262 Parse date strings in vorbis comments that have an invalid (zero)
22263 month or day (#410396).
22264 * tests/check/libs/tag.c: (GST_START_TEST):
22265 Test case for the above.
22267 2007-02-24 20:12:49 +0000 Loïc Minier <lool+gnome@via.ecp.fr>
22269 Fix compilation with LDFLAGS='-Wl,-z,defs' (#410963).
22270 Original commit message from CVS:
22271 Patch by: Loïc Minier <lool+gnome at via ecp fr>
22273 * ext/alsa/Makefile.am:
22274 * gst/audiotestsrc/Makefile.am:
22275 Fix compilation with LDFLAGS='-Wl,-z,defs' (#410963).
22277 2007-02-23 18:49:29 +0000 Tim-Philipp Müller <tim@centricular.net>
22279 gst/playback/gstplaybin.c: Improve docs: point out that the application needs to assist playbin with buffering.
22280 Original commit message from CVS:
22281 * gst/playback/gstplaybin.c:
22282 Improve docs: point out that the application needs to assist playbin
22285 2007-02-23 13:10:50 +0000 Tim-Philipp Müller <tim@centricular.net>
22287 Change GStreamer marker prefix in detail string from 'gstreamer.net' to just 'gstreamer'. Document the caps string co...
22288 Original commit message from CVS:
22289 * gst-libs/gst/utils/install-plugins.c:
22290 * gst-libs/gst/utils/missing-plugins.c:
22291 * tests/check/libs/utils.c: (missing_msg_check_getters):
22292 Change GStreamer marker prefix in detail string from 'gstreamer.net'
22293 to just 'gstreamer'. Document the caps string component of the
22294 decoder/encoder detail a bit better, since not everyone will be
22295 familiar with the GStreamer media type/caps system (but they better
22296 enjoy nested itemized lists).
22298 2007-02-22 12:57:47 +0000 Tim-Philipp Müller <tim@centricular.net>
22300 gst-libs/gst/netbuffer/gstnetbuffer.c: Fix copying of GstNetBuffer (would crash before, or at least lead to invalid m...
22301 Original commit message from CVS:
22302 * gst-libs/gst/netbuffer/gstnetbuffer.c:
22303 (notgst_buffer_copy_fields_in_place), (gst_netbuffer_copy):
22304 Fix copying of GstNetBuffer (would crash before, or at least lead to
22305 invalid memory access, #410772), for now by copying the GstBuffer copy
22306 code from the core over here so we can copy the GstBuffer fields on a
22307 provided buffer instance (of type GstNetBuffer in this case). Would be
22308 better to fix this with some support by the core though (and in the long
22309 run change the broken GstBuffer/GstMiniObject copy semantics, #393099).
22310 * tests/check/Makefile.am:
22311 Enable unit test for GstNetBuffer.
22313 2007-02-22 11:04:10 +0000 Andy Wingo <wingo@pobox.com>
22316 * gst-libs/gst/audio/gstbaseaudiosink.c:
22317 gst-libs/gst/audio/gstbaseaudiosink.c
22318 Original commit message from CVS:
22319 2007-02-22 Andy Wingo <wingo@pobox.com>
22320 * gst-libs/gst/audio/gstbaseaudiosink.c
22321 (gst_base_audio_sink_init): Disable pull-mode activation until we
22322 figure out how to make audio sinks go to PLAYING.
22324 2007-02-22 09:04:37 +0000 Stefan Kost <ensonic@users.sourceforge.net>
22326 Add float as an intermediate format, as well as float mixing. Enable test that was failing before. Fixes #339837
22327 Original commit message from CVS:
22328 * gst/audioconvert/audioconvert.c: (float), (double), (float_hq),
22329 (double_hq), (audio_convert_get_func_index),
22330 (audio_convert_prepare_context), (audio_convert_convert):
22331 * gst/audioconvert/audioconvert.h:
22332 * gst/audioconvert/gstchannelmix.c: (gst_channel_mix_setup_matrix),
22333 (gst_channel_mix_mix_int), (gst_channel_mix_mix_float):
22334 * gst/audioconvert/gstchannelmix.h:
22335 * tests/check/elements/audioconvert.c: (GST_START_TEST):
22336 Add float as an intermediate format, as well as float mixing. Enable
22337 test that was failing before. Fixes #339837
22339 2007-02-21 16:12:15 +0000 Jan Schmidt <thaytan@mad.scientist.com>
22341 tests/examples/seek/seek.c: Undo the previous commit: -1 as a stop time implies that the stop time is the end of file...
22342 Original commit message from CVS:
22343 * tests/examples/seek/seek.c: (do_seek):
22344 Undo the previous commit: -1 as a stop time implies that the stop
22345 time is the end of file, clearing any previously configured segment.
22347 2007-02-21 15:36:26 +0000 Jan Schmidt <thaytan@mad.scientist.com>
22349 tests/examples/seek/seek.c: Don't SEEK_SET with a stop time of -1, use SEEK_NONE instead.
22350 Original commit message from CVS:
22351 * tests/examples/seek/seek.c: (do_seek):
22352 Don't SEEK_SET with a stop time of -1, use SEEK_NONE instead.
22354 2007-02-21 13:55:54 +0000 Stefan Kost <ensonic@users.sourceforge.net>
22356 gst/volume/gstvolume.c: Unbreak volume, value remains gint.
22357 Original commit message from CVS:
22358 * gst/volume/gstvolume.c: (volume_process_int16),
22359 (volume_process_int16_clamp), (volume_set_caps):
22360 Unbreak volume, value remains gint.
22362 2007-02-21 13:08:51 +0000 Stefan Kost <ensonic@users.sourceforge.net>
22364 gst/volume/gstvolume.*: Extend float audio support (double) and some int->uint cleanups.
22365 Original commit message from CVS:
22366 * gst/volume/gstvolume.c: (volume_choose_func),
22367 (volume_update_real_volume), (gst_volume_set_volume),
22368 (gst_volume_init), (volume_process_double), (volume_process_float),
22369 (volume_process_int16), (volume_process_int16_clamp),
22370 (volume_set_caps), (volume_transform_ip), (volume_update_volume):
22371 * gst/volume/gstvolume.h:
22372 Extend float audio support (double) and some int->uint cleanups.
22374 2007-02-20 15:44:32 +0000 Edward Hervey <bilboed@bilboed.com>
22376 gst/playback/gstdecodebin2.c: Don't free groups from the streaming threads. Just put them aside and free them in disp...
22377 Original commit message from CVS:
22378 * gst/playback/gstdecodebin2.c: (gst_decode_bin_dispose),
22379 (multi_queue_underrun_cb), (gst_decode_group_check_if_drained),
22380 (sort_end_pads), (gst_decode_group_expose),
22381 (gst_decode_group_hide):
22382 Don't free groups from the streaming threads. Just put them aside and
22383 free them in dispose.
22385 2007-02-20 11:20:52 +0000 Edward Hervey <bilboed@bilboed.com>
22387 gst/playback/gstdecodebin2.c: Handle dynamic pads within groups.
22388 Original commit message from CVS:
22389 * gst/playback/gstdecodebin2.c: (connect_element),
22390 (pad_added_group_cb), (gst_decode_group_check_if_blocked),
22391 (sort_end_pads), (gst_decode_group_expose):
22392 Handle dynamic pads within groups.
22393 Sort pads before exposing them in order to make playbin happy.
22394 There still is a race with the multiqueue filling up. This should be
22398 2007-02-18 21:02:36 +0000 Tim-Philipp Müller <tim@centricular.net>
22400 gst-libs/gst/utils/: Some more docs (and descriptions for two subtitle formats).
22401 Original commit message from CVS:
22402 * gst-libs/gst/utils/base-utils.c:
22403 * gst-libs/gst/utils/descriptions.c:
22404 * gst-libs/gst/utils/install-plugins.c:
22405 * gst-libs/gst/utils/missing-plugins.c:
22406 Some more docs (and descriptions for two subtitle formats).
22408 2007-02-16 10:19:45 +0000 Tim-Philipp Müller <tim@centricular.net>
22410 gst-libs/gst/audio/audio.c: Fix documentation.
22411 Original commit message from CVS:
22412 * gst-libs/gst/audio/audio.c:
22415 2007-02-16 10:15:46 +0000 Yves Lefebvre <ivanohe@abacom.com>
22417 gst/videorate/gstvideorate.c: Don't leak caps. Fixes #408278.
22418 Original commit message from CVS:
22419 Patch by: Yves Lefebvre <ivanohe abacom com>
22420 * gst/videorate/gstvideorate.c: (gst_video_rate_setcaps):
22421 Don't leak caps. Fixes #408278.
22423 2007-02-15 15:17:23 +0000 Stefan Kost <ensonic@users.sourceforge.net>
22425 More docs coverage and some ChangeLog surgery (add missing names)
22426 Original commit message from CVS:
22427 * ext/cdparanoia/gstcdparanoiasrc.h:
22428 * ext/ogg/gstoggdemux.h:
22429 * gst-libs/gst/audio/audio.c: (gst_audio_frame_byte_size),
22430 (gst_audio_frame_length), (gst_audio_duration_from_pad_buffer),
22431 (gst_audio_is_buffer_framed), (gst_audio_structure_set_int):
22432 * gst-libs/gst/audio/audio.h:
22433 * gst-libs/gst/audio/gstaudiofilter.h:
22434 * gst-libs/gst/interfaces/videoorientation.h:
22435 * gst/adder/gstadder.h:
22436 More docs coverage and some ChangeLog surgery (add missing names)
22438 2007-02-15 12:07:57 +0000 Wim Taymans <wim.taymans@gmail.com>
22440 sys/: Small constifications.
22441 Original commit message from CVS:
22442 * sys/ximage/ximagesink.c:
22443 (gst_ximagesink_calculate_pixel_aspect_ratio):
22444 * sys/xvimage/xvimagesink.c:
22445 (gst_xvimagesink_calculate_pixel_aspect_ratio):
22446 Small constifications.
22448 2007-02-15 12:06:25 +0000 Wim Taymans <wim.taymans@gmail.com>
22450 gst-libs/gst/audio/gstbaseaudiosink.c: Answer latency query.
22451 Original commit message from CVS:
22452 * gst-libs/gst/audio/gstbaseaudiosink.c:
22453 (gst_base_audio_sink_class_init), (gst_base_audio_sink_query),
22454 (gst_base_audio_sink_render), (gst_base_audio_sink_callback),
22455 (gst_base_audio_sink_async_play),
22456 (gst_base_audio_sink_change_state):
22457 Answer latency query.
22458 Use configured latency when syncing.
22460 * gst-libs/gst/audio/gstbaseaudiosrc.c:
22461 (gst_base_audio_src_class_init), (gst_base_audio_src_dispose),
22462 (gst_base_audio_src_query), (gst_base_audio_src_change_state):
22463 Fix possible memleak.
22464 Implement latency query.
22467 2007-02-15 11:59:41 +0000 Wim Taymans <wim.taymans@gmail.com>
22469 ext/alsa/gstalsasink.c: Ignore errors in reset, these are not fatal. They also grab the element lock which is already...
22470 Original commit message from CVS:
22471 * ext/alsa/gstalsasink.c: (gst_alsasink_reset):
22472 Ignore errors in reset, these are not fatal. They also grab the element
22473 lock which is already taking when this function is called. Fixes
22476 2007-02-13 13:50:56 +0000 Christian Schaller <uraeus@gnome.org>
22478 * gst-plugins-base.spec.in:
22479 add header file for easy codec install
22480 Original commit message from CVS:
22481 add header file for easy codec install
22483 2007-02-13 10:24:13 +0000 Stefan Kost <ensonic@users.sourceforge.net>
22485 configure.ac: Remove 'tests/examples/xerror/Makefile' from output files again.
22486 Original commit message from CVS:
22488 Remove 'tests/examples/xerror/Makefile' from output files again.
22490 2007-02-13 09:12:11 +0000 Stefan Kost <ensonic@users.sourceforge.net>
22492 Also crossref against gst-plugins-base-libs.
22493 Original commit message from CVS:
22495 * docs/plugins/Makefile.am:
22496 Also crossref against gst-plugins-base-libs.
22498 2007-02-12 20:42:23 +0000 Stefan Kost <ensonic@users.sourceforge.net>
22500 Add crossreferences to glib/gobject/gstream docs.
22501 Original commit message from CVS:
22503 * docs/libs/Makefile.am:
22504 * docs/plugins/Makefile.am:
22505 Add crossreferences to glib/gobject/gstream docs.
22506 * gst-libs/gst/audio/audio.h:
22508 * gst/audiotestsrc/gstaudiotestsrc.c: (plugin_init):
22509 Add own debug category.
22511 2007-02-12 11:01:04 +0000 René Stadler <mail@renestadler.de>
22513 gst-libs/gst/tag/gstvorbistag.c: Add vorbis/FLAC-tag mapping for new GST_TAG_REFERENCE_LEVEL (#403597).
22514 Original commit message from CVS:
22515 Patch by: René Stadler <mail at renestadler de>
22516 * gst-libs/gst/tag/gstvorbistag.c:
22517 Add vorbis/FLAC-tag mapping for new GST_TAG_REFERENCE_LEVEL
22520 2007-02-12 10:33:40 +0000 Tim-Philipp Müller <tim@centricular.net>
22522 gst/playback/gstplaybasebin.c: When we have external subtitles and wait for the subtitle decodebin to get up and runn...
22523 Original commit message from CVS:
22524 * gst/playback/gstplaybasebin.c: (setup_source):
22525 When we have external subtitles and wait for the subtitle decodebin
22526 to get up and running, we set up a (sync) bus handler for the
22527 subtitle decodebin, so we can stop waiting when it posts an error
22528 message. However, we should do that before we set the subtitle
22529 decodebin's state to playing, otherwise things are racy and we might
22530 miss error messages posted before we had a chance to set up the bus.
22531 This should finally fix totem hanging on .txt pseudo-subtitle files.
22533 2007-02-10 19:27:48 +0000 Sébastien Moutte <sebastien@moutte.net>
22535 gst-libs/gst/rtp/gstbasertpaudiopayload.c: Use gst_gdouble_to_guint64 for conversions.
22536 Original commit message from CVS:
22537 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:(gst_base_rtp_audio_payload_handle_frame_based_buffer):
22538 Use gst_gdouble_to_guint64 for conversions.
22539 * win32/common/config.h.in:
22540 Add a define for GST_INSTALL_PLUGINS_HELPER
22541 * win32/common/libgstaudio.def:
22542 * win32/common/libgstcdda.def:
22543 * win32/common/libgstnetbuffer.def:
22544 * win32/common/libgstrtp.def:
22545 * win32/common/libgutils.def:
22546 Add new exported functions.
22547 * win32/vs6/gst_plugins_base.dsw:
22548 * win32/vs6/libgstdecodebin.dsp:
22549 * win32/vs6/libgstnetbuffer.dsp:
22550 * win32/vs6/libgstplaybin.dsp:
22551 * win32/vs6/libgstrtp.dsp:
22552 * win32/vs6/libgstvorbis.dsp:
22553 * win32/vs6/libgstcdda.dsp:
22554 * win32/vs6/libgstgdp.dsp:
22555 * win32/vs6/libgstutils.dsp:
22556 Update and add new project files.
22558 2007-02-10 18:19:37 +0000 Tim-Philipp Müller <tim@centricular.net>
22560 gst/subparse/gstsubparse.c: For SubRip (.srt) subtitles, ignore all markup tags we don't handle (like font tags, for ...
22561 Original commit message from CVS:
22562 * gst/subparse/gstsubparse.c: (subrip_remove_unhandled_tag),
22563 (subrip_remove_unhandled_tags), (parse_subrip):
22564 For SubRip (.srt) subtitles, ignore all markup tags we don't
22565 handle (like font tags, for example).
22566 * tests/check/elements/subparse.c:
22569 2007-02-09 13:28:01 +0000 Tim-Philipp Müller <tim@centricular.net>
22573 Original commit message from CVS:
22576 2007-02-09 13:16:27 +0000 Tim-Philipp Müller <tim@centricular.net>
22578 gst/playback/: Don't error out if there is no fakesink in the READY to NULL state change, since when decodebin is re-...
22579 Original commit message from CVS:
22580 * gst/playback/gstdecodebin.c: (add_fakesink),
22581 (gst_decode_bin_change_state):
22582 * gst/playback/gstdecodebin2.c: (add_fakesink),
22583 (gst_decode_bin_change_state):
22584 Don't error out if there is no fakesink in the READY to NULL state
22585 change, since when decodebin is re-used, we're only adding the
22586 fakesink element in READY to PAUSED.
22587 * tests/check/elements/decodebin.c:
22588 (new_decoded_pad_plug_fakesink_cb), (GST_START_TEST),
22590 Minimal unit test to make sure we can use the same decodebin
22591 instance twice (at least with audiotestsrc input).
22593 2007-02-09 09:58:28 +0000 Tim-Philipp Müller <tim@centricular.net>
22595 ext/alsa/gstalsa.c: Try to get devic-name from device string first, and from handle only as fallback (seems to yield ...
22596 Original commit message from CVS:
22597 * ext/alsa/gstalsa.c: (gst_alsa_find_device_name):
22598 Try to get devic-name from device string first, and from handle only
22599 as fallback (seems to yield better results and is more robust
22600 against buggy probing code on the application side).
22602 2007-02-08 15:43:26 +0000 Julien Puydt <julien.puydt@laposte.net>
22604 ext/alsa/: Improve device-name detection a bit, especially in the case where the device is not actually open (#405020...
22605 Original commit message from CVS:
22606 Based on patch by: Julien Puydt <julien.puydt at laposte net>
22607 * ext/alsa/gstalsa.c: (gst_alsa_find_device_name_no_handle),
22608 (gst_alsa_find_device_name):
22609 * ext/alsa/gstalsa.h:
22610 * ext/alsa/gstalsasink.c: (gst_alsasink_get_property):
22611 * ext/alsa/gstalsasrc.c: (gst_alsasrc_get_property):
22612 Improve device-name detection a bit, especially in the case where
22613 the device is not actually open (#405020, #405024). Move common code
22614 into gstalsa.c instead of duplicating it.
22616 2007-02-07 13:05:01 +0000 Tim-Philipp Müller <tim@centricular.net>
22618 gst/audioconvert/gstaudioconvert.c: Fix up docs chunk so that gtk-doc doesn't complain, and fix typo.
22619 Original commit message from CVS:
22620 * gst/audioconvert/gstaudioconvert.c:
22621 Fix up docs chunk so that gtk-doc doesn't complain, and fix typo.
22623 2007-02-06 17:47:32 +0000 Julien Moutte <julien@moutte.net>
22625 sys/xvimage/xvimagesink.*: Implement PropertyProbe Interface for XVAdaptors so that one can choose the adaptor to use...
22626 Original commit message from CVS:
22627 2007-02-06 Julien MOUTTE <julien@moutte.net>
22628 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_handle_xevents),
22629 (gst_xvimagesink_get_xv_support),
22630 (gst_xvimagesink_xcontext_clear),
22631 (gst_xvimagesink_interface_supported),
22632 (gst_xvimagesink_probe_get_properties),
22633 (gst_xvimagesink_probe_probe_property),
22634 (gst_xvimagesink_probe_needs_probe),
22635 (gst_xvimagesink_probe_get_values),
22636 (gst_xvimagesink_property_probe_interface_init),
22637 (gst_xvimagesink_set_property), (gst_xvimagesink_get_property),
22638 (gst_xvimagesink_init), (gst_xvimagesink_class_init),
22639 (gst_xvimagesink_get_type):
22640 * sys/xvimage/xvimagesink.h: Implement PropertyProbe Interface
22641 for XVAdaptors so that one can choose the adaptor to use with
22642 gstreamer-properties.
22644 2007-02-06 14:00:31 +0000 Stefan Kost <ensonic@users.sourceforge.net>
22646 gst/audioconvert/gstaudioconvert.c: Also mention that a conversion from double to float is suboptimal still.
22647 Original commit message from CVS:
22648 * gst/audioconvert/gstaudioconvert.c:
22649 Also mention that a conversion from double to float is suboptimal still.
22651 2007-02-06 09:42:05 +0000 Tim-Philipp Müller <tim@centricular.net>
22653 gst-libs/gst/audio/gstaudiofilter.c: Clear our formats structure and free the caps contained in it when shutting down.
22654 Original commit message from CVS:
22655 * gst-libs/gst/audio/gstaudiofilter.c:
22656 (gst_audio_filter_class_init), (gst_audio_filter_change_state):
22657 Clear our formats structure and free the caps contained in it when
22660 2007-02-05 18:39:51 +0000 Andy Wingo <wingo@pobox.com>
22663 * gst-libs/gst/audio/gstbaseaudiosink.c:
22664 gst-libs/gst/audio/gstbaseaudiosink.c
22665 Original commit message from CVS:
22666 2007-02-05 Andy Wingo <wingo@pobox.com>
22667 * gst-libs/gst/audio/gstbaseaudiosink.c
22668 (gst_base_audio_sink_callback): Update basesink->offset so that we
22669 pull monotonically increasing offsets instead of, um, seeking back
22670 to 0 each time. Fixes alsasrc ! alsasink!
22672 2007-02-05 11:44:52 +0000 Tim-Philipp Müller <tim@centricular.net>
22674 gst/videoscale/gstvideoscale.c: A width and height of 1 makes us crash, so increase minimum size to 2x2 pixels until ...
22675 Original commit message from CVS:
22676 * gst/videoscale/gstvideoscale.c:
22677 A width and height of 1 makes us crash, so increase minimum size to
22678 2x2 pixels until someone feels like fixing this (#404512).
22680 2007-02-04 16:23:37 +0000 Tim-Philipp Müller <tim@centricular.net>
22682 tests/check/pipelines/oggmux.c: Add small test to make sure request pads are cleaned up properly even if oggmux never...
22683 Original commit message from CVS:
22684 * tests/check/pipelines/oggmux.c: (GST_START_TEST), (oggmux_suite):
22685 Add small test to make sure request pads are cleaned up properly
22686 even if oggmux never changes state out of NULL.
22688 2007-02-04 14:11:51 +0000 Tim-Philipp Müller <tim@centricular.net>
22690 tests/check/libs/utils.c: Fix unit test. Turns out things work much better when you
22691 Original commit message from CVS:
22692 * tests/check/libs/utils.c: (GST_START_TEST):
22693 Fix unit test. Turns out things work much better when you
22694 NULL-terminate string arrays. Should make p5 build bot happy again.
22696 2007-02-03 23:28:45 +0000 Tim-Philipp Müller <tim@centricular.net>
22698 gst-libs/gst/audio/: Oops, forgot to commit fixed-up example.
22699 Original commit message from CVS:
22700 * gst-libs/gst/audio/Makefile.am:
22701 * gst-libs/gst/audio/gstaudiofiltertemplate.c:
22702 (gst_audio_filter_template_base_init),
22703 (gst_audio_filter_template_class_init),
22704 (gst_audio_filter_template_init),
22705 (gst_audio_filter_template_set_property),
22706 (gst_audio_filter_template_get_property),
22707 (gst_audio_filter_template_setup),
22708 (gst_audio_filter_template_filter),
22709 (gst_audio_filter_template_filter_inplace), (plugin_init):
22710 Oops, forgot to commit fixed-up example.
22712 2007-02-03 20:19:35 +0000 Tim-Philipp Müller <tim@centricular.net>
22714 Port GstAudioFilter to 0.10. This change technically breaks but seems justifiable on the grounds that the base class ...
22715 Original commit message from CVS:
22716 * docs/libs/gst-plugins-base-libs-sections.txt:
22717 * gst-libs/gst/audio/gstaudiofilter.c: (gst_audio_filter_get_type),
22718 (gst_audio_filter_class_init), (gst_audio_filter_init),
22719 (gst_audio_filter_set_caps),
22720 (gst_audio_filter_class_add_pad_templates):
22721 * gst-libs/gst/audio/gstaudiofilter.h:
22722 Port GstAudioFilter to 0.10. This change technically breaks
22723 API and ABI (and thus also every library developer's heart),
22724 but seems justifiable on the grounds that the base class was
22725 completely unusable before (ie. would crash immediately when
22726 actually used). Fixes #403963 (and eventually also #403572).
22727 Also document all of this a bit.
22729 2007-02-03 14:26:54 +0000 Tim-Philipp Müller <tim@centricular.net>
22731 Lowering log level to see why things fail on the p5 build bot; fix some typos in unit test messages.
22732 Original commit message from CVS:
22733 * gst-libs/gst/utils/install-plugins.c:
22734 (gst_install_plugins_spawn_child):
22735 * tests/check/libs/utils.c:
22736 (test_base_utils_install_plugins_do_callout):
22737 Lowering log level to see why things fail on the p5 build bot;
22738 fix some typos in unit test messages.
22740 2007-02-03 13:59:27 +0000 Tim-Philipp Müller <tim@centricular.net>
22742 tests/check/libs/utils.c: Don't hard-code temp directory for test helper; use GLib functions to write out file and do...
22743 Original commit message from CVS:
22744 * tests/check/libs/utils.c:
22745 (test_base_utils_install_plugins_do_callout):
22746 Don't hard-code temp directory for test helper; use GLib functions
22747 to write out file and do error checking etc.
22749 2007-02-02 20:42:08 +0000 Tim-Philipp Müller <tim@centricular.net>
22751 gst-libs/gst/utils/: API: add API for applications to initiate installation of missing plugins, ie. gst_install_plugi...
22752 Original commit message from CVS:
22753 * gst-libs/gst/utils/Makefile.am:
22754 * gst-libs/gst/utils/base-utils.h:
22755 * gst-libs/gst/utils/install-plugins.c:
22756 (gst_install_plugins_context_set_xid),
22757 (gst_install_plugins_context_new),
22758 (gst_install_plugins_context_free),
22759 (gst_install_plugins_get_helper),
22760 (gst_install_plugins_spawn_child),
22761 (gst_install_plugins_return_from_status),
22762 (gst_install_plugins_installer_exited),
22763 (gst_install_plugins_async), (gst_install_plugins_sync),
22764 (gst_install_plugins_return_get_name),
22765 (gst_install_plugins_installation_in_progress):
22766 * gst-libs/gst/utils/install-plugins.h:
22767 API: add API for applications to initiate installation of missing
22768 plugins, ie. gst_install_plugins_async() primarily.
22769 Based on libgimme-codec by Ryan Lortie.
22771 Add --with-install-plugins-helper configure option so distros can specify
22772 the path of the helper script or program to call when plugin installation
22773 is requested (distros: please do any argument munging in this helper
22774 script instead of patching GStreamer to pass arguments differently
22775 to another program directly).
22776 * docs/libs/gst-plugins-base-libs-docs.sgml:
22777 * docs/libs/gst-plugins-base-libs-sections.txt:
22778 Build and document new API.
22779 * tests/check/libs/utils.c: (result_cb),
22780 (test_base_utils_install_plugins_do_callout), (GST_START_TEST),
22781 (libgstbaseutils_suite):
22782 Some simple checks for the new API.
22784 2007-02-02 14:44:29 +0000 Tim-Philipp Müller <tim@centricular.net>
22786 tests/check/elements/audioconvert.c: Add small test for 32bit float <=> 64bit float conversion (works only one way so...
22787 Original commit message from CVS:
22788 * tests/check/elements/audioconvert.c: (test_float_conversion):
22789 Add small test for 32bit float <=> 64bit float conversion (works
22790 only one way so far, 32=>64 produces structured noise).
22792 2007-02-02 11:21:48 +0000 Tim-Philipp Müller <tim@centricular.net>
22794 gst/audioconvert/gstaudioconvert.c: We don't support floats with a width of 40, 48 or 56 bits.
22795 Original commit message from CVS:
22796 * gst/audioconvert/gstaudioconvert.c:
22797 (set_structure_widths_32_and_64), (make_lossless_changes):
22798 We don't support floats with a width of 40, 48 or 56 bits.
22800 2007-02-02 09:48:53 +0000 Stefan Kost <ensonic@users.sourceforge.net>
22802 gst/audioconvert/: Support for 64-bit float audio in audioconvert (#339837)
22803 Original commit message from CVS:
22804 * gst/audioconvert/audioconvert.c: (float), (double),
22805 (audio_convert_get_func_index):
22806 * gst/audioconvert/gstaudioconvert.c: (set_structure_widths),
22807 (make_lossless_changes):
22808 Support for 64-bit float audio in audioconvert (#339837)
22810 2007-02-01 18:50:08 +0000 Holger Wansing <linux@wansing-online.de>
22812 po/: Add German translation (#352069).
22813 Original commit message from CVS:
22814 Patch by: Holger Wansing <linux wansing-online de>
22817 Add German translation (#352069).
22819 2007-02-01 17:52:39 +0000 Sebastian Dröge <slomo@circular-chaos.org>
22821 ext/ogg/gstoggmux.c: Use newly added GstCollectPads API to free the allocated resources in the GstOggPad structures (...
22822 Original commit message from CVS:
22823 reviewed by: Wim Taymans <wim@fluendo.com>
22824 * ext/ogg/gstoggmux.c: (gst_ogg_mux_ogg_pad_destroy_notify),
22825 (gst_ogg_mux_request_new_pad), (gst_ogg_mux_release_pad):
22826 Use newly added GstCollectPads API to free the allocated resources in
22827 the GstOggPad structures (#402393).
22829 2007-01-31 15:58:53 +0000 Jan Schmidt <thaytan@mad.scientist.com>
22831 gst/playback/gstplaybin.c: Add audioresample+audioconvert in front of the visualisation element, so that elements lik...
22832 Original commit message from CVS:
22833 * gst/playback/gstplaybin.c: (gen_vis_element):
22834 Add audioresample+audioconvert in front of the visualisation
22835 element, so that elements like libvisual 0.4 that don't support all
22836 samplerates can work.
22839 2007-01-30 19:19:37 +0000 Tim-Philipp Müller <tim@centricular.net>
22841 gst/playback/gstplaybasebin.c: Take some locks and make a copy of the streaminfo value array we maintain while holdin...
22842 Original commit message from CVS:
22843 * gst/playback/gstplaybasebin.c: (gst_play_base_bin_get_property),
22844 (gst_play_base_bin_get_streaminfo_value_array):
22845 Take some locks and make a copy of the streaminfo value array we
22846 maintain while holding the lock, so that the application can
22847 retrieve the stream-info as a value array in a thread-safe way.
22849 2007-01-30 11:29:17 +0000 Wim Taymans <wim.taymans@gmail.com>
22851 gst/audioconvert/gstaudioconvert.c: Don't fail on 0 sized buffers. Fixes #396835.
22852 Original commit message from CVS:
22853 * gst/audioconvert/gstaudioconvert.c:
22854 Don't fail on 0 sized buffers. Fixes #396835.
22856 2007-01-29 21:13:07 +0000 David Schleef <ds@schleef.org>
22858 gst/typefind/gsttypefindfunctions.c: Detect BBCD as video/x-dirac, so we can play raw dirac streams.
22859 Original commit message from CVS:
22860 * gst/typefind/gsttypefindfunctions.c:
22861 Detect BBCD as video/x-dirac, so we can play raw dirac
22864 2007-01-29 18:14:25 +0000 Tim-Philipp Müller <tim@centricular.net>
22866 ext/theora/theoraenc.c: Check return value of theora_encode_header(), or we might try to allocate a random number of ...
22867 Original commit message from CVS:
22868 * ext/theora/theoraenc.c: (theora_enc_chain):
22869 Check return value of theora_encode_header(), or we might try to
22870 allocate a random number of bytes. theora_encode_header() can fail
22871 if libtheora has been compiled with encoding support disabled.
22874 2007-01-29 10:53:06 +0000 Wim Taymans <wim.taymans@gmail.com>
22876 tests/check/gst/.cvsignore: Do as buildbot says.
22877 Original commit message from CVS:
22878 * tests/check/gst/.cvsignore:
22879 Do as buildbot says.
22881 2007-01-29 10:25:11 +0000 Wim Taymans <wim.taymans@gmail.com>
22883 ext/libvisual/visual.c: Fix strides in libvisual. Gst uses X strides.
22884 Original commit message from CVS:
22885 * ext/libvisual/visual.c: (gst_visual_src_setcaps):
22886 Fix strides in libvisual. Gst uses X strides.
22887 Inspired by: <ed at catmur dot co dot uk> and
22888 <tim at centricular dot net>
22891 2007-01-27 13:32:24 +0000 Wim Taymans <wim.taymans@gmail.com>
22893 ext/ogg/gstoggdemux.*: Properly propagate streaming errors when we are scanning the file for chains so that we don't ...
22894 Original commit message from CVS:
22895 * ext/ogg/gstoggdemux.c: (gst_ogg_demux_submit_buffer),
22896 (gst_ogg_demux_get_data), (gst_ogg_demux_get_next_page),
22897 (gst_ogg_demux_get_prev_page), (gst_ogg_demux_do_seek),
22898 (gst_ogg_demux_perform_seek),
22899 (gst_ogg_demux_bisect_forward_serialno),
22900 (gst_ogg_demux_read_chain), (gst_ogg_demux_read_end_chain),
22901 (gst_ogg_demux_find_chains), (gst_ogg_demux_handle_page),
22902 (gst_ogg_demux_chain), (gst_ogg_demux_combine_flows),
22903 (gst_ogg_demux_loop_reverse), (gst_ogg_demux_loop):
22904 * ext/ogg/gstoggdemux.h:
22905 Properly propagate streaming errors when we are scanning the file for
22906 chains so that we don't crash when shut down. Might fix some crashers
22907 when quickly switching oggs in RB such as #332503 and #378436.
22909 2007-01-26 12:44:46 +0000 Tim-Philipp Müller <tim@centricular.net>
22911 ext/gnomevfs/gstgnomevfssrc.c: Map a gnome-vfs HOST_NOT_FOUND error into a GStreamer NOT_FOUND error code as well.
22912 Original commit message from CVS:
22913 * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_start):
22914 Map a gnome-vfs HOST_NOT_FOUND error into a GStreamer NOT_FOUND
22915 error code as well.
22917 2007-01-25 16:02:41 +0000 Wim Taymans <wim.taymans@gmail.com>
22919 gst/playback/gstplaybasebin.c: Don't try to disconnect a signal from a finalized object.
22920 Original commit message from CVS:
22921 * gst/playback/gstplaybasebin.c: (remove_source):
22922 Don't try to disconnect a signal from a finalized object.
22924 2007-01-25 14:29:21 +0000 Tim-Philipp Müller <tim@centricular.net>
22926 gst/playback/gstdecodebin2.c: Cast lock macro parameters to make sure we're actually accessing the lock member at the...
22927 Original commit message from CVS:
22928 * gst/playback/gstdecodebin2.c: (gst_decode_bin_dispose):
22929 Cast lock macro parameters to make sure we're actually accessing the
22930 lock member at the right class level. Free list itself in _dispose()
22931 as well and NULL it in case dispose gets called multiple times.
22933 2007-01-25 14:02:37 +0000 Edward Hervey <bilboed@bilboed.com>
22935 gst/playback/gstdecodebin2.c: Free GstDecodeGroups no longer used.
22936 Original commit message from CVS:
22937 * gst/playback/gstdecodebin2.c:
22938 (gst_decode_bin_dispose),(gst_decode_bin_finalize):
22939 Free GstDecodeGroups no longer used.
22940 (gst_decode_group_expose):
22941 Don't unlock too many times !
22942 (deactivate_free_recursive):
22943 Free iterator once we're done with it.
22944 Fix for recursively deactivating elements (stop at ghostpads).
22946 2007-01-25 12:24:18 +0000 Tim-Philipp Müller <tim@centricular.net>
22948 gst/playback/gstplaybin.c: Fix up caps on the frame buffer before we save it and potentially make it accessible to ot...
22949 Original commit message from CVS:
22950 * gst/playback/gstplaybin.c: (handoff):
22951 Fix up caps on the frame buffer before we save it and potentially
22952 make it accessible to other threads via g_object_get; also use
22953 gst_buffer_replace() instead of gst_mini_object_replace().
22955 2007-01-25 12:06:59 +0000 Tim-Philipp Müller <tim@centricular.net>
22957 gst/playback/gstplaybin.c: Make getting the current frame thread-safe.
22958 Original commit message from CVS:
22959 * gst/playback/gstplaybin.c: (gst_play_bin_get_property):
22960 Make getting the current frame thread-safe.
22962 2007-01-25 11:48:10 +0000 Edward Hervey <bilboed@bilboed.com>
22964 gst/playback/gstdecodebin2.c: Set queues to bigger sizes to cope with HD contents.
22965 Original commit message from CVS:
22966 * gst/playback/gstdecodebin2.c: (gst_decode_bin_finalize),
22967 (gst_decode_group_new), (gst_decode_group_free):
22968 Set queues to bigger sizes to cope with HD contents.
22969 Fix some mutex freeing and add comment about MT safe methods.
22971 2007-01-24 12:51:20 +0000 Tim-Philipp Müller <tim@centricular.net>
22973 ext/pango/gsttextoverlay.c: Don't unnecessarily ref (and then leak) upstream events if the text pad is not linked. Fi...
22974 Original commit message from CVS:
22975 * ext/pango/gsttextoverlay.c: (gst_text_overlay_src_event),
22976 (gst_text_overlay_text_event):
22977 Don't unnecessarily ref (and then leak) upstream events if the text
22978 pad is not linked. Fixes #399948.
22979 * tests/check/gst-plugins-base.supp:
22980 Add suppression for pango on edgy/x86 for textoverlay test.
22982 2007-01-24 12:10:56 +0000 Wim Taymans <wim.taymans@gmail.com>
22984 gst-libs/gst/rtp/gstrtpbuffer.h: Add some more fixed payloads.
22985 Original commit message from CVS:
22986 * gst-libs/gst/rtp/gstrtpbuffer.h:
22987 Add some more fixed payloads.
22989 2007-01-23 18:39:45 +0000 Tim-Philipp Müller <tim@centricular.net>
22991 ext/ogg/gstoggdemux.c: Error out properly if we get an error from libogg while reading the
22992 Original commit message from CVS:
22993 * ext/ogg/gstoggdemux.c: (gst_ogg_demux_read_chain):
22994 Error out properly if we get an error from libogg while reading the
22995 BOS page(s). Fixes crash parsing 'fuzzed' ogg file (#399340).
22997 2007-01-23 17:49:29 +0000 Tim-Philipp Müller <tim@centricular.net>
22999 gst/playback/gstdecodebin2.c: Don't leak mutex.
23000 Original commit message from CVS:
23001 * gst/playback/gstdecodebin2.c: (gst_decode_bin_finalize):
23003 * tests/check/elements/playbin.c:
23004 (test_sink_usage_video_only_stream),
23005 (test_suburi_error_unknowntype), (test_suburi_error_invalidfile),
23006 (test_suburi_error_wrongproto), (test_missing_urisource_handler),
23007 (test_missing_suburisource_handler),
23008 (test_missing_primary_decoder), (playbin_suite):
23009 Run all tests once with decodebin and once with decodebin2.
23010 One test does not pass yet with decodebin2.
23012 2007-01-23 14:30:28 +0000 Edward Hervey <bilboed@bilboed.com>
23014 ext/ogg/gstoggmux.c: Fix the cases where oggmux doesn't properly figure out that all sinkpads have gone EOS, and ther...
23015 Original commit message from CVS:
23016 * ext/ogg/gstoggmux.c: (all_pads_eos), (gst_ogg_mux_collected):
23017 Fix the cases where oggmux doesn't properly figure out that all
23018 sinkpads have gone EOS, and therefore doesn't push out the remaining
23019 buffers and the final EOS event.
23022 2007-01-23 13:19:19 +0000 Julien Moutte <julien@moutte.net>
23024 sys/: Don't lock on navigation event push, just on keysym to string.
23025 Original commit message from CVS:
23026 2007-01-23 Julien MOUTTE <julien@moutte.net>
23027 * sys/ximage/ximagesink.c: (gst_ximagesink_handle_xevents):
23028 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_handle_xevents):
23029 Don't lock on navigation event push, just on keysym to string.
23030 Fixes #397673 again.
23032 2007-01-22 17:37:38 +0000 Edward Hervey <bilboed@bilboed.com>
23034 gst/playback/gstdecodebin2.c: Cleanups.
23035 Original commit message from CVS:
23036 * gst/playback/gstdecodebin2.c: (gst_decode_group_new),
23037 (get_current_group), (group_demuxer_event_probe),
23038 (gst_decode_group_expose), (deactivate_free_recursive),
23039 (gst_decode_group_free):
23041 Don't forget to emit 'no-more-pads' once a group is exposed.
23042 Cleanup elements from a DecodeGroup once we remove it.
23043 Protect call to gst_decode_group_expose() with the decodebin lock.
23045 2007-01-22 13:16:42 +0000 Julien Moutte <julien@moutte.net>
23047 sys/: Looking at Xorg code i can't figure out if that XKeysymToString function is thread sensible or not. Lock it jus...
23048 Original commit message from CVS:
23049 2007-01-22 Julien MOUTTE <julien@moutte.net>
23050 * sys/ximage/ximagesink.c: (gst_ximagesink_handle_xevents):
23051 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_handle_xevents):
23052 Looking at Xorg code i can't figure out if that XKeysymToString
23053 function is thread sensible or not. Lock it just in case as
23054 recommended by Radek Doulik <rodo at ximian dot com>.
23056 2007-01-22 13:10:13 +0000 Julien Moutte <julien@moutte.net>
23058 sys/: Lock that X Call as well. Fixes #397673.
23059 Original commit message from CVS:
23060 2007-01-22 Julien MOUTTE <julien@moutte.net>
23061 * sys/ximage/ximagesink.c: (gst_ximagesink_handle_xevents):
23062 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_handle_xevents):
23063 Lock that X Call as well. Fixes #397673.
23065 2007-01-22 12:03:27 +0000 Tim-Philipp Müller <tim@centricular.net>
23067 gst/typefind/gsttypefindfunctions.c: Don't go into an endless loop if the file starts with 00 00 01 2X, like quicktim...
23068 Original commit message from CVS:
23069 * gst/typefind/gsttypefindfunctions.c: (mpeg4_video_type_find):
23070 Don't go into an endless loop if the file starts with 00 00 01 2X,
23071 like quicktime redirect files might. Fixes #396042.
23072 * tests/check/Makefile.am:
23073 * tests/check/gst/.cvsignore:
23074 * tests/check/gst/typefindfunctions.c: (GST_START_TEST),
23075 (typefindfunctions_suite):
23076 Add unit test for the above.
23078 2007-01-22 10:27:26 +0000 Tim-Philipp Müller <tim@centricular.net>
23080 gst-libs/gst/riff/riff-media.c: On second thought, use "depth" field rather than "bpp" field.
23081 Original commit message from CVS:
23082 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
23083 On second thought, use "depth" field rather than "bpp" field.
23085 2007-01-22 09:23:01 +0000 Tim-Philipp Müller <tim@centricular.net>
23087 gst-libs/gst/riff/riff-media.c: Camtasia caps apparently need a bpp field (#398875).
23088 Original commit message from CVS:
23089 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
23090 Camtasia caps apparently need a bpp field (#398875).
23092 2007-01-19 19:09:05 +0000 Tim-Philipp Müller <tim@centricular.net>
23094 gst/playback/gstplaybasebin.c: Attempt at a better error message in case we don't have the required
23095 Original commit message from CVS:
23096 * gst/playback/gstplaybasebin.c: (setup_subtitle),
23097 (gen_source_element), (gst_play_base_bin_change_state):
23098 Attempt at a better error message in case we don't have the required
23099 URI handler installed; post missing-plugin message also when we're
23100 missing an URI handler for the subtitle URI; clean up properly also
23101 when an error occurs and we never made it to PAUSED state.
23102 * tests/check/elements/playbin.c: (GST_START_TEST),
23104 Check that we're also getting a missing-plugin messsage for a
23105 missing subtitle URI handler (and clean up properly).
23107 2007-01-19 18:47:30 +0000 Tim-Philipp Müller <tim@centricular.net>
23109 gst/playback/gstplaybasebin.c: Plug a few reference leaks.
23110 Original commit message from CVS:
23111 * gst/playback/gstplaybasebin.c: (analyse_source), (setup_source):
23112 Plug a few reference leaks.
23114 2007-01-19 12:23:06 +0000 Tim-Philipp Müller <tim@centricular.net>
23116 gst/typefind/gsttypefindfunctions.c: Lower probability a bit if the marker isn't right at the start, to decrease the ...
23117 Original commit message from CVS:
23118 * gst/typefind/gsttypefindfunctions.c: (mpeg2_sys_type_find):
23119 Lower probability a bit if the marker isn't right at the start,
23120 to decrease the chance of false positives.
23122 2007-01-19 11:31:50 +0000 Tim-Philipp Müller <tim@centricular.net>
23124 gst/typefind/gsttypefindfunctions.c: Small mpeg2 system stream typefinding improvement: make typefinder probe a bit i...
23125 Original commit message from CVS:
23126 * gst/typefind/gsttypefindfunctions.c: (mpeg2_sys_type_find):
23127 Small mpeg2 system stream typefinding improvement: make typefinder
23128 probe a bit into the stream instead of just looking for a marker
23129 at the beginning. Fixes #397810.
23131 2007-01-18 16:23:35 +0000 Tim-Philipp Müller <tim@centricular.net>
23133 gst/audioconvert/gstchannelmix.c: Remove compatibility cruft for prehistoric GLib versions.
23134 Original commit message from CVS:
23135 * gst/audioconvert/gstchannelmix.c:
23136 Remove compatibility cruft for prehistoric GLib versions.
23138 2007-01-17 16:11:14 +0000 Tim-Philipp Müller <tim@centricular.net>
23140 gst/playback/: Let decodebin be the element to post missing-plugin messages for missing decoders (rather than playbin...
23141 Original commit message from CVS:
23142 * gst/playback/Makefile.am:
23143 * gst/playback/gstdecodebin.c: (close_pad_link):
23144 * gst/playback/gstdecodebin2.c: (analyze_new_pad):
23145 * gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init),
23146 (gst_play_base_bin_handle_message_func), (unknown_type):
23147 Let decodebin be the element to post missing-plugin messages for
23148 missing decoders (rather than playbin); make playbin implement
23149 GstBin::handle_message so we can suppress missing-plugin messages
23150 for types we're not handling on purpose (don't want to bring up an
23151 installer in those cases).
23153 2007-01-16 19:37:55 +0000 Tim-Philipp Müller <tim@centricular.net>
23155 gst/: Fix potentially unaligned access (#397207).
23156 Original commit message from CVS:
23157 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
23158 * gst-libs/gst/tag/gstvorbistag.c:
23159 (gst_tag_list_to_vorbiscomment_buffer):
23160 * gst/typefind/gsttypefindfunctions.c: (vorbis_type_find):
23161 Fix potentially unaligned access (#397207).
23163 2007-01-16 12:17:06 +0000 Stefan Kost <ensonic@users.sourceforge.net>
23165 tests/examples/seek/seek.c: Allow to toggle looping while it plays. Fix callback prototype. Clean up code a bit more....
23166 Original commit message from CVS:
23167 * tests/examples/seek/seek.c: (set_scale), (update_scale),
23168 (do_seek), (stop_seek), (pause_cb), (stop_cb), (loop_toggle_cb),
23169 (rate_spinbutton_changed_cb), (msg_eos), (msg_segment_done),
23171 Allow to toggle looping while it plays. Fix callback prototype. Clean
23172 up code a bit more. Add copyright header.
23174 2007-01-16 11:41:58 +0000 Stefan Kost <ensonic@users.sourceforge.net>
23176 sys/xvimage/xvimagesink.c: Red and blue mask was swapped (spotted by Dan Williams).
23177 Original commit message from CVS:
23178 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support):
23179 Red and blue mask was swapped (spotted by Dan Williams).
23181 2007-01-15 13:58:58 +0000 Stefan Kost <ensonic@users.sourceforge.net>
23183 gst-libs/gst/tag/: Use new beats-per-minute tag from core.
23184 Original commit message from CVS:
23185 * gst-libs/gst/tag/gstid3tag.c:
23186 * gst-libs/gst/tag/gstvorbistag.c:
23187 Use new beats-per-minute tag from core.
23189 2007-01-15 11:30:53 +0000 Tim-Philipp Müller <tim@centricular.net>
23191 po/POTFILES.in: Add new files with translatable strings, so they actually make it into the template file one day.
23192 Original commit message from CVS:
23194 Add new files with translatable strings, so they actually make it
23195 into the template file one day.
23197 2007-01-12 21:19:35 +0000 Andy Wingo <wingo@pobox.com>
23200 * gst-libs/gst/audio/gstbaseaudiosink.c:
23201 * gst-libs/gst/audio/gstbaseaudiosrc.c:
23202 gst-libs/gst/audio/gstbaseaudiosink.c
23203 Original commit message from CVS:
23204 2007-01-12 Andy Wingo <wingo@pobox.com>
23205 * gst-libs/gst/audio/gstbaseaudiosink.c
23206 (gst_base_audio_sink_fixate): Implement, stolen from baseaudiosrc.
23207 (gst_base_audio_sink_activate_pull): Remove the handwavey nego
23208 stuff, as the base class handles this now. Actually tell the ring
23210 (gst_base_audio_sink_callback): Cast the ring buffer correctly.
23211 How did this work before? Maybe I'm not as awesome a programmer as
23213 * gst-libs/gst/audio/gstbaseaudiosrc.c
23214 (gst_base_audio_src_fixate): Rework as a basesrc vmethod instead
23217 2007-01-12 18:08:23 +0000 Tim-Philipp Müller <tim@centricular.net>
23219 gst-libs/gst/utils/missing-plugins.c: Remove more fields so that the application can better blacklist formats that ha...
23220 Original commit message from CVS:
23221 * gst-libs/gst/utils/missing-plugins.c: (copy_and_clean_caps):
23222 Remove more fields so that the application can better blacklist
23223 formats that have been tried before.
23225 2007-01-12 17:43:40 +0000 Christian Schaller <uraeus@gnome.org>
23227 * gst-plugins-base.spec.in:
23229 Original commit message from CVS:
23232 2007-01-12 12:47:29 +0000 Tim-Philipp Müller <tim@centricular.net>
23234 gst-libs/gst/audio/mixerutils.h: Add G_BEGIN_DECLS and G_END_DECLS guards so these helpers can be used when compiling...
23235 Original commit message from CVS:
23236 * gst-libs/gst/audio/mixerutils.h:
23237 Add G_BEGIN_DECLS and G_END_DECLS guards so these helpers can be
23238 used when compiling with c++ compilers as well.
23240 2007-01-12 09:45:23 +0000 Tim-Philipp Müller <tim@centricular.net>
23242 gst/typefind/gsttypefindfunctions.c: Fix comment.
23243 Original commit message from CVS:
23244 * gst/typefind/gsttypefindfunctions.c:
23247 2007-01-11 13:12:17 +0000 Tim-Philipp Müller <tim@centricular.net>
23249 gst/playback/gstplaybin.c: Post missing-plugin messages also when we error out because converters, textoverlay or aut...
23250 Original commit message from CVS:
23251 * gst/playback/gstplaybin.c: (post_missing_element_message),
23252 (gen_video_element), (gen_text_element), (gen_audio_element),
23254 Post missing-plugin messages also when we error out because
23255 converters, textoverlay or auto*sinks are missing (#161922).
23257 2007-01-10 16:08:18 +0000 Wim Taymans <wim.taymans@gmail.com>
23259 gst/playback/: Fix the case where we try to ref a NULL element when we delay a link because of unfixed caps.
23260 Original commit message from CVS:
23261 * gst/playback/gstdecodebin.c: (dynamic_add), (close_pad_link),
23262 (is_demuxer_element), (new_caps):
23263 * gst/playback/gstplaybasebin.c: (source_new_pad):
23264 Fix the case where we try to ref a NULL element when we delay a link
23265 because of unfixed caps.
23266 Set the state of autoplugged decodebins to PAUSED.
23267 RTSP now works in playbin, we can remove it from the blacklist.
23269 2007-01-09 14:33:24 +0000 Tim-Philipp Müller <tim@centricular.net>
23271 gst/playback/: Post missing-plugin messages on the bus for missing sources and missing decoders/demuxers/depayloaders...
23272 Original commit message from CVS:
23273 * gst/playback/Makefile.am:
23274 * gst/playback/gstplaybasebin.c: (string_arr_has_str),
23275 (unknown_type), (setup_subtitle), (gen_source_element):
23276 * gst/playback/gstplaybin.c: (plugin_init):
23277 Post missing-plugin messages on the bus for missing sources and
23278 missing decoders/demuxers/depayloaders; fix error code used when
23279 we're missing an URI handler source; for media types that we are not
23280 handling on purpose at the moment, don't print "don't know how to
23281 handle xyz" messages to the terminal or post missing-plugin
23282 messages on the bus.
23283 * tests/check/elements/playbin.c: (create_playbin),
23284 (GST_START_TEST), (gst_codec_src_uri_get_type),
23285 (gst_codec_src_uri_get_protocols), (gst_codec_src_uri_get_uri),
23286 (gst_codec_src_uri_set_uri), (gst_codec_src_uri_handler_init),
23287 (gst_codec_src_init_type), (gst_codec_src_base_init),
23288 (gst_codec_src_create), (gst_codec_src_class_init),
23289 (gst_codec_src_init), (plugin_init), (playbin_suite):
23290 Add some tests for the missing-plugin stuff.
23292 2007-01-09 14:20:08 +0000 Tim-Philipp Müller <tim@centricular.net>
23294 API: add new libgstbaseutils library with functions
23295 Original commit message from CVS:
23297 * gst-libs/gst/Makefile.am:
23298 * gst-libs/gst/utils/Makefile.am:
23299 * gst-libs/gst/utils/base-utils.c: (gst_base_utils_init):
23300 * gst-libs/gst/utils/base-utils.h:
23301 * gst-libs/gst/utils/descriptions.c: (format_info_get_desc),
23302 (find_format_info), (caps_are_rtp_caps),
23303 (gst_base_utils_get_source_description),
23304 (gst_base_utils_get_sink_description),
23305 (gst_base_utils_get_decoder_description),
23306 (gst_base_utils_get_encoder_description),
23307 (gst_base_utils_get_element_description),
23308 (gst_base_utils_add_codec_description_to_tag_list),
23309 (gst_base_utils_get_codec_description), (gst_base_utils_list_all):
23310 * gst-libs/gst/utils/descriptions.h:
23311 * gst-libs/gst/utils/missing-plugins.c:
23312 (missing_structure_get_type), (copy_and_clean_caps),
23313 (gst_missing_uri_source_message_new),
23314 (gst_missing_uri_sink_message_new),
23315 (gst_missing_element_message_new),
23316 (gst_missing_decoder_message_new),
23317 (gst_missing_encoder_message_new),
23318 (missing_structure_get_string_detail),
23319 (missing_structure_get_caps_detail),
23320 (gst_missing_plugin_message_get_installer_detail),
23321 (gst_missing_plugin_message_get_description),
23322 (gst_is_missing_plugin_message):
23323 * gst-libs/gst/utils/missing-plugins.h:
23324 API: add new libgstbaseutils library with functions
23325 - to create and parse missing-plugins messages
23326 - that provide (translated) descriptions for caps/decoders/sources/etc.
23328 * pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
23329 * pkgconfig/gstreamer-plugins-base.pc.in:
23331 * docs/libs/gst-plugins-base-libs-docs.sgml:
23332 * docs/libs/gst-plugins-base-libs-sections.txt:
23333 Generate docs for new lib and API.
23334 * tests/check/Makefile.am:
23335 * tests/check/libs/.cvsignore:
23336 * tests/check/libs/utils.c: (missing_msg_check_getters),
23337 (GST_START_TEST), (libgstbaseutils_suite):
23338 Add some basic unit tests.
23340 2007-01-09 13:35:08 +0000 Tim-Philipp Müller <tim@centricular.net>
23342 ext/ogg/Makefile.am: Dist gstoggdemux.h to fix 'make distcheck'.
23343 Original commit message from CVS:
23344 * ext/ogg/Makefile.am:
23345 Dist gstoggdemux.h to fix 'make distcheck'.
23346 * sys/v4l/Makefile.am:
23347 Fix 'make distcheck' even more.
23349 2007-01-09 12:30:46 +0000 Wim Taymans <wim.taymans@gmail.com>
23352 Original commit message from CVS:
23353 * docs/plugins/Makefile.am:
23354 * docs/plugins/gst-plugins-base-plugins-docs.sgml:
23355 * docs/plugins/gst-plugins-base-plugins-sections.txt:
23356 * ext/ogg/gstoggdemux.c: (gst_ogg_page_copy), (gst_ogg_page_free),
23357 (gst_ogg_pad_query_types), (gst_ogg_pad_submit_page),
23358 (gst_ogg_chain_reset), (gst_ogg_chain_new_stream),
23359 (gst_ogg_demux_perform_seek):
23360 * ext/ogg/gstoggdemux.h:
23362 Add some more comments.
23365 2007-01-09 11:15:57 +0000 Wim Taymans <wim.taymans@gmail.com>
23367 Small documentation updates/fixes
23368 Original commit message from CVS:
23369 * ext/theora/theoradec.c:
23370 * ext/vorbis/vorbisdec.c:
23371 * gst-libs/gst/audio/gstringbuffer.c:
23372 (gst_ring_buffer_commit_full):
23373 * gst-libs/gst/audio/gstringbuffer.h:
23374 * gst-libs/gst/rtp/gstrtpbuffer.c:
23375 * gst-libs/gst/tag/gstvorbistag.c:
23376 Small documentation updates/fixes
23378 2007-01-09 10:37:01 +0000 Tim-Philipp Müller <tim@centricular.net>
23380 configure.ac: Require core CVS HEAD for Andy's basesrc/sink API additions.
23381 Original commit message from CVS:
23383 Require core CVS HEAD for Andy's basesrc/sink API additions.
23385 2007-01-08 14:01:23 +0000 Günter Thelen <daedalus.inc@gmx.net>
23387 gst/typefind/gsttypefindfunctions.c: Add typefinder for flac-in-ogg in conformance with the ogg-mapping on flac.sf.ne...
23388 Original commit message from CVS:
23389 Patch by: Günter Thelen <daedalus dot inc at gmx net>
23390 * gst/typefind/gsttypefindfunctions.c: (flac_type_find),
23392 Add typefinder for flac-in-ogg in conformance with the ogg-mapping
23393 on flac.sf.net (there appear to be other versions of the first
23394 ogg page in the wild) (#391365).
23396 2007-01-08 13:32:32 +0000 Tim-Philipp Müller <tim@centricular.net>
23398 configure.ac: Check if localtime_r() is available.
23399 Original commit message from CVS:
23401 Check if localtime_r() is available.
23402 * ext/pango/gstclockoverlay.c: (gst_clock_overlay_render_time):
23403 If localtime_r() is not available, fall back to localtime(). Should
23404 fix build on MingW (#393310).
23406 2007-01-08 12:30:03 +0000 Tim-Philipp Müller <tim@centricular.net>
23408 gst/subparse/gstsubparse.*: Remove spurious 1000 subtrahend when calculating the timestamp from the frame number and ...
23409 Original commit message from CVS:
23410 * gst/subparse/gstsubparse.c: (parse_mdvdsub):
23411 * gst/subparse/gstsubparse.h:
23412 Remove spurious 1000 subtrahend when calculating the timestamp from
23413 the frame number and the frame rate . Also, use the frames/second
23414 value specified in the first line of the file, if one is specified
23415 there. Should fix #357503.
23416 * tests/check/elements/subparse.c: (do_test),
23417 (test_tmplayer_do_test), (test_microdvd_do_test), (GST_START_TEST),
23419 Add some basic unit tests for the microdvd subtitle format.
23421 2007-01-07 21:53:38 +0000 Young-Ho Cha <ganadist@chollian.net>
23423 sys/xvimage/xvimagesink.c: Fixes : #390076.
23424 Original commit message from CVS:
23425 2007-01-07 Julien MOUTTE <julien@moutte.net>
23426 * sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_destroy),
23427 (gst_xvimage_buffer_finalize), (gst_xvimagesink_xvimage_new),
23428 (gst_xvimagesink_xvimage_put),
23429 (gst_lookup_xv_port_from_adaptor),
23430 (gst_xvimagesink_get_xv_support), (gst_xvimagesink_setcaps),
23431 (gst_xvimagesink_set_xwindow_id),
23432 (gst_xvimagesink_set_event_handling),
23433 (gst_xvimagesink_set_property), (gst_xvimagesink_get_property),
23434 (gst_xvimagesink_init), (gst_xvimagesink_class_init):
23435 Patch by : Young-Ho Cha <ganadist at chollian dot net>
23437 Add an adaptor property to select a specific XV adaptor.
23438 * sys/xvimage/xvimagesink.h:
23440 2007-01-07 18:50:13 +0000 Julien Moutte <julien@moutte.net>
23442 sys/: Use flow_lock much more to protect every access to xwindow.
23443 Original commit message from CVS:
23444 2007-01-07 Julien MOUTTE <julien@moutte.net>
23445 * sys/ximage/ximagesink.c: (gst_ximage_buffer_finalize),
23446 (gst_ximagesink_handle_xerror), (gst_ximagesink_ximage_new),
23447 (gst_ximagesink_ximage_destroy), (gst_ximagesink_ximage_put),
23448 (gst_ximagesink_handle_xevents), (gst_ximagesink_setcaps),
23449 (gst_ximagesink_change_state), (gst_ximagesink_set_xwindow_id),
23450 (gst_ximagesink_expose), (gst_ximagesink_set_event_handling):
23451 * sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_destroy),
23452 (gst_xvimage_buffer_finalize), (gst_xvimagesink_handle_xerror),
23453 (gst_xvimagesink_xvimage_new), (gst_xvimagesink_xvimage_put),
23454 (gst_xvimagesink_handle_xevents), (gst_xvimagesink_setcaps),
23455 (gst_xvimagesink_change_state),
23456 (gst_xvimagesink_set_xwindow_id),
23457 (gst_xvimagesink_expose), (gst_xvimagesink_set_event_handling):
23458 Use flow_lock much more to protect every access to xwindow.
23459 Try to catch erros while creating images in case some drivers
23461 just generating an XError when the requested image is too big.
23462 Should fix : #354698, #384008, #384060.
23463 * tests/icles/stress-xoverlay.c: (cycle_window),
23465 Implement some stress testing of setting window xid.
23467 2007-01-07 10:33:55 +0000 Sébastien Moutte <sebastien@moutte.net>
23469 win32/common/libgsaudio.def: Add new exported function.
23470 Original commit message from CVS:
23471 * win32/common/libgsaudio.def:
23472 Add new exported function.
23473 * win32/common/libgstogg.dsp:
23474 Add gstoggaviparse.c to the build.
23475 * win32/common/libgstvideoscale.dsp:
23476 Add vs_4tap.c to the build.
23477 * win32/common/libgstvorbis.dsp:
23478 Add vorbistag.c to the build.
23480 2007-01-06 17:28:40 +0000 Andy Wingo <wingo@pobox.com>
23483 * gst-libs/gst/audio/gstbaseaudiosink.c:
23484 gst-libs/gst/audio/gstbaseaudiosink.c (gst_base_audio_sink_class_init)
23485 Original commit message from CVS:
23486 2007-01-06 Andy Wingo <wingo@pobox.com>
23487 * gst-libs/gst/audio/gstbaseaudiosink.c
23488 (gst_base_audio_sink_class_init)
23489 (gst_base_audio_sink_init):
23490 (gst_base_audio_sink_activate_pull): Add an activate_pull function
23491 to baseaudiosink, and tell basesink that we can work in pull mode.
23492 This way the ring buffer thread drives the pipeline directly, if
23493 pull mode is possible. There is some lingering nastiness regarding
23495 (gst_base_audio_sink_callback): Implement the callback to pull
23496 data. This interface is a bit light, though -- it should get a
23497 GstFlowReturn return value at least.
23499 2007-01-05 19:43:55 +0000 Tim-Philipp Müller <tim@centricular.net>
23501 Printf format and missing argument fixes.
23502 Original commit message from CVS:
23503 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_stream_out):
23504 * ext/vorbis/vorbisdec.c: (vorbis_handle_data_packet):
23505 * gst/playback/gstdecodebin2.c:
23506 (gst_decode_group_check_if_blocked):
23507 Printf format and missing argument fixes.
23509 2007-01-05 18:57:53 +0000 Jan Schmidt <thaytan@mad.scientist.com>
23511 ext/ogg/gstogmparse.c: Activate pads before adding them to the element.
23512 Original commit message from CVS:
23513 * ext/ogg/gstogmparse.c: (gst_ogm_parse_stream_header),
23514 (gst_ogm_parse_change_state):
23515 Activate pads before adding them to the element.
23517 2007-01-05 16:02:50 +0000 Tim-Philipp Müller <tim@centricular.net>
23519 tests/examples/seek/: Call g_thread_init() first thing in main() (see #391278).
23520 Original commit message from CVS:
23521 * tests/examples/seek/scrubby.c: (main):
23522 * tests/examples/seek/seek.c: (main):
23523 Call g_thread_init() first thing in main() (see #391278).
23525 2007-01-05 12:19:34 +0000 Tim-Philipp Müller <tim@centricular.net>
23527 tests/check/: Add test for GstNetBuffer + gst_buffer_copy(). Disabled for the time being, since it's broken, see #393...
23528 Original commit message from CVS:
23529 * tests/check/Makefile.am:
23530 * tests/check/libs/.cvsignore:
23531 * tests/check/libs/netbuffer.c: (GST_START_TEST),
23533 Add test for GstNetBuffer + gst_buffer_copy(). Disabled
23534 for the time being, since it's broken, see #393099.
23536 2007-01-05 12:13:24 +0000 Tim-Philipp Müller <tim@centricular.net>
23538 tests/check/Makefile.am: Update to use GST_PLUGINS_BASE_CFLAGS as well.
23539 Original commit message from CVS:
23540 * tests/check/Makefile.am:
23541 Update to use GST_PLUGINS_BASE_CFLAGS as well.
23543 2007-01-04 12:49:48 +0000 Thomas Vander Stichele <thomas@apestaart.org>
23545 configure.ac: split out GST_CFLAGS into GST_PLUGINS_BASE_CFLAGS and GST_CFLAGS so that GST_BASE_CFLAGS can go inbetwe...
23546 Original commit message from CVS:
23548 split out GST_CFLAGS into GST_PLUGINS_BASE_CFLAGS and GST_CFLAGS
23549 so that GST_BASE_CFLAGS can go inbetween them, making sure
23550 we use uninstalled gst-libs headers
23551 * docs/libs/Makefile.am:
23552 * ext/alsa/Makefile.am:
23553 * ext/cdparanoia/Makefile.am:
23554 * ext/gnomevfs/Makefile.am:
23555 * ext/libvisual/Makefile.am:
23556 * ext/ogg/Makefile.am:
23557 * ext/theora/Makefile.am:
23558 * ext/vorbis/Makefile.am:
23559 * gst-libs/gst/audio/Makefile.am:
23560 * gst-libs/gst/cdda/Makefile.am:
23561 * gst-libs/gst/interfaces/Makefile.am:
23562 * gst-libs/gst/riff/Makefile.am:
23563 * gst-libs/gst/rtp/Makefile.am:
23564 * gst-libs/gst/tag/Makefile.am:
23565 * gst/adder/Makefile.am:
23566 * gst/audioconvert/Makefile.am:
23567 * gst/audiorate/Makefile.am:
23568 * gst/audioresample/Makefile.am:
23569 * gst/playback/Makefile.am:
23570 * gst/tcp/Makefile.am:
23571 * gst/videoscale/Makefile.am:
23572 * gst/volume/Makefile.am:
23573 * sys/ximage/Makefile.am:
23574 * sys/xvimage/Makefile.am:
23575 * tests/icles/Makefile.am:
23578 2007-01-04 11:30:53 +0000 Julien Moutte <julien@moutte.net>
23580 Add a method to the XOverlay interface to allow disabling of event handling in x[v]imagesink elements. This will let ...
23581 Original commit message from CVS:
23582 2007-01-04 Julien MOUTTE <julien@moutte.net>
23583 * gst-libs/gst/interfaces/xoverlay.c:
23584 (gst_x_overlay_handle_events):
23585 * gst-libs/gst/interfaces/xoverlay.h:
23586 * sys/ximage/ximagesink.c: (gst_ximagesink_xwindow_new),
23587 (gst_ximagesink_set_xwindow_id),
23588 (gst_ximagesink_set_event_handling),
23589 (gst_ximagesink_xoverlay_init), (gst_ximagesink_set_property),
23590 (gst_ximagesink_get_property), (gst_ximagesink_init),
23591 (gst_ximagesink_class_init):
23592 * sys/ximage/ximagesink.h:
23593 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xwindow_new),
23594 (gst_xvimagesink_set_xwindow_id),
23595 (gst_xvimagesink_set_event_handling),
23596 (gst_xvimagesink_xoverlay_init), (gst_xvimagesink_set_property),
23597 (gst_xvimagesink_get_property), (gst_xvimagesink_init),
23598 (gst_xvimagesink_class_init):
23599 * sys/xvimage/xvimagesink.h:
23600 * tests/icles/stress-xoverlay.c: (toggle_events),
23602 Add a method to the XOverlay interface to allow disabling of
23603 event handling in x[v]imagesink elements. This will let X events
23604 propagate to parent windows which can be usefull in some cases.
23605 Be carefull that the application is then responsible of pushing
23606 navigation events and expose events to the video sink.
23609 2007-01-03 15:45:06 +0000 Tim-Philipp Müller <tim@centricular.net>
23611 Add vorbistag <=> GStreamer tag mapping for GST_TAG_LOCATION (fixes #392070).
23612 Original commit message from CVS:
23613 * gst-libs/gst/tag/gstvorbistag.c:
23614 * tests/check/libs/tag.c: (GST_START_TEST):
23615 Add vorbistag <=> GStreamer tag mapping for GST_TAG_LOCATION
23618 2007-01-01 18:59:49 +0000 Tim-Philipp Müller <tim@centricular.net>
23621 Original commit message from CVS:
23623 * docs/Makefile.am:
23624 * docs/design/Makefile.am:
23627 2006-12-27 17:15:35 +0000 Julien Moutte <julien@moutte.net>
23629 docs/libs/gst-plugins-base-libs-sections.txt: Fix a documentation typo. Fixes: #390063.
23630 Original commit message from CVS:
23631 2006-12-27 Julien MOUTTE <julien@moutte.net>
23632 * docs/libs/gst-plugins-base-libs-sections.txt: Fix a
23634 typo. Fixes: #390063.
23636 2006-12-27 12:08:13 +0000 Julien Moutte <julien@moutte.net>
23638 sys/: Plug a caps leak.
23639 Original commit message from CVS:
23640 2006-12-27 Julien MOUTTE <julien@moutte.net>
23641 * sys/ximage/ximagesink.c: (gst_ximagesink_setcaps):
23642 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_setcaps): Plug a
23644 * win32/common/config.h: Updated.
23646 2006-12-22 12:10:18 +0000 Stefan Kost <ensonic@users.sourceforge.net>
23648 tests/check/elements/: Fix the dp tests, but activating the pads for the streamheader tests too and cleaning up condi...
23649 Original commit message from CVS:
23650 * tests/check/elements/gdpdepay.c: (cleanup_gdpdepay),
23651 (setup_gdpdepay_streamheader):
23652 * tests/check/elements/gdppay.c: (cleanup_gdppay),
23653 (setup_gdppay_streamheader):
23654 Fix the dp tests, but activating the pads for the streamheader tests
23655 too and cleaning up conditionaly
23657 2006-12-22 11:09:34 +0000 Jan Schmidt <thaytan@mad.scientist.com>
23659 gst/ffmpegcolorspace/: Add 2 new caps arrangements, for 24-bit RGB and BGR in 32-bits, but at the other end of the wo...
23660 Original commit message from CVS:
23661 * gst/ffmpegcolorspace/avcodec.h:
23662 * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
23663 (gst_ffmpeg_pixfmt_to_caps), (gst_ffmpeg_caps_to_pixfmt),
23664 (gst_ffmpegcsp_avpicture_fill):
23665 * gst/ffmpegcolorspace/imgconvert.c: (img_convert),
23666 (img_get_alpha_info):
23667 Add 2 new caps arrangements, for 24-bit RGB and BGR in 32-bits, but at the
23668 other end of the word. Fixes: #387073.
23669 Add some inconsequential branch hints in a couple of places.
23671 2006-12-21 12:30:11 +0000 Tim-Philipp Müller <tim@centricular.net>
23673 gst/ffmpegcolorspace/gstffmpegcodecmap.c: The "signed" field in raw audio caps is of boolean type, trying to extract ...
23674 Original commit message from CVS:
23675 * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
23676 (gst_ffmpeg_caps_to_smpfmt):
23677 The "signed" field in raw audio caps is of boolean type, trying to
23678 extract the value with _get_int() will fail (fix to keep in sync with
23679 the copy in gst-ffmpeg)
23681 2006-12-21 08:12:26 +0000 Stefan Kost <ensonic@users.sourceforge.net>
23683 tests/check/elements/: consistent pad (de)activation
23684 Original commit message from CVS:
23685 * tests/check/elements/audioresample.c: (cleanup_audioresample):
23686 * tests/check/elements/audiotestsrc.c: (cleanup_audiotestsrc):
23687 * tests/check/elements/gdpdepay.c: (setup_gdpdepay),
23688 (cleanup_gdpdepay):
23689 * tests/check/elements/gdppay.c: (setup_gdppay), (cleanup_gdppay):
23690 * tests/check/elements/subparse.c: (teardown_subparse):
23691 * tests/check/elements/textoverlay.c: (cleanup_textoverlay):
23692 * tests/check/elements/videorate.c: (cleanup_videorate):
23693 * tests/check/elements/videotestsrc.c: (cleanup_videotestsrc):
23694 * tests/check/elements/volume.c: (cleanup_volume):
23695 * tests/check/elements/vorbisdec.c: (setup_vorbisdec),
23696 (cleanup_vorbisdec):
23697 * tests/check/elements/vorbistag.c: (setup_vorbistag),
23698 (cleanup_vorbistag):
23699 consistent pad (de)activation
23701 2006-12-20 10:29:58 +0000 Tim-Philipp Müller <tim@centricular.net>
23703 gst/typefind/gsttypefindfunctions.c: Forgot to register the extensions.
23704 Original commit message from CVS:
23705 * gst/typefind/gsttypefindfunctions.c: (plugin_init):
23706 Forgot to register the extensions.
23708 2006-12-20 09:25:55 +0000 Tim-Philipp Müller <tim@centricular.net>
23710 gst/typefind/gsttypefindfunctions.c: Add typefinder for VIVO files (my christmas present to the 90s).
23711 Original commit message from CVS:
23712 * gst/typefind/gsttypefindfunctions.c: (vivo_type_find),
23714 Add typefinder for VIVO files (my christmas present to the 90s).
23716 2006-12-16 13:59:09 +0000 Tim-Philipp Müller <tim@centricular.net>
23718 gst/playback/gstdecodebin.c: Special-case the text/plain media type: we only want to recognise it as a 'raw' decoded ...
23719 Original commit message from CVS:
23720 * gst/playback/gstdecodebin.c: (type_found):
23721 Special-case the text/plain media type: we only want to recognise it
23722 as a 'raw' decoded media type if it comes from a demuxer or subtitle
23723 parser, but not if the entire stream is of text/plain type. If the
23724 entire stream is text/plain, we should just error out.
23725 This fixes playback of audio files with lyrics in totem. Totem can't
23726 distinguish between text files and subtitle files and passes any
23727 .txt file with the same basename as the main file to playbin as
23728 suburi, and playbin will then throw a 'subtitle found, but no video
23729 stream' error, which isn't entirely helpful. See #380342.
23730 Also, with this change we'll show a slightly more correct error
23731 message in case totem passes a playlist file to us (although a
23732 custom error message wording instead of the default text would
23733 probably not be a bad idea either).
23734 Same problem also needs to be fixed for playbin+decodebin2.
23735 * tests/check/Makefile.am:
23736 * tests/check/elements/decodebin.c: (src_handoff_cb),
23737 (decodebin_new_decoded_pad_cb), (GST_START_TEST),
23739 Add simple unit test for decodebin for the above.
23741 2006-12-16 12:22:57 +0000 Tim-Philipp Müller <tim@centricular.net>
23743 gst/playback/: Refuse to change state to READY when we failed to create any of the required elements in our instance ...
23744 Original commit message from CVS:
23745 * gst/playback/gstdecodebin.c: (gst_decode_bin_change_state):
23746 * gst/playback/gstdecodebin2.c: (gst_decode_bin_change_state):
23747 Refuse to change state to READY when we failed to create any of the
23748 required elements in our instance init function.
23750 2006-12-15 10:52:23 +0000 Tim-Philipp Müller <tim@centricular.net>
23752 docs/libs/gst-plugins-base-libs-sections.txt: Small docs fixes/updates.
23753 Original commit message from CVS:
23754 * docs/libs/gst-plugins-base-libs-sections.txt:
23755 Small docs fixes/updates.
23756 * gst-libs/gst/video/gstvideosink.h:
23757 Remove nonfunctional GST_VIDEO_SINK_CLOCK macro which is a leftover
23758 from the 0.9 days (GST_BASE_SINK_CLOCK, which it points to, was
23759 removed from the base sink API between 0.9.6 and 0.9.7).
23760 API: add GST_VIDEO_SINK_CAST and use it for the height/width
23761 accessor macros, so we don't do a runtime GObject type check every
23764 2006-12-15 00:20:37 +0000 Thomas Vander Stichele <thomas@apestaart.org>
23767 Original commit message from CVS:
23769 * gst-plugins-base.doap:
23770 * gst-plugins-base.spec.in:
23773 2006-12-09 15:12:38 +0000 Jens Granseuer <jensgr@gmx.net>
23775 Declare variables at the beginning of a block. Fixes #383195.
23776 Original commit message from CVS:
23777 Patch by: Jens Granseuer <jensgr at gmx net>
23778 * gst-libs/gst/cdda/gstcddabasesrc.c: (gst_cdda_base_src_create):
23779 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
23780 (gst_base_rtp_audio_payload_handle_frame_based_buffer),
23781 (gst_base_rtp_audio_payload_handle_sample_based_buffer):
23782 * sys/v4l/gstv4lsrc.c: (gst_v4lsrc_fixate):
23783 Declare variables at the beginning of a block. Fixes #383195.
23785 2006-12-07 02:38:41 +0000 Jan Schmidt <thaytan@mad.scientist.com>
23787 configure.ac: Bump version nano - back to CVS.
23788 Original commit message from CVS:
23790 Bump version nano - back to CVS.
23792 === release 0.10.11 ===
23794 2006-12-07 02:30:38 +0000 Jan Schmidt <thaytan@mad.scientist.com>
23796 configure.ac: releasing 0.10.11, "Dumb things"
23797 Original commit message from CVS:
23798 === release 0.10.11 ===
23799 2006-12-06 Jan Schmidt <thaytan@mad.scientist.com>
23801 releasing 0.10.11, "Dumb things"
23803 2006-12-05 12:44:00 +0000 Jan Schmidt <thaytan@mad.scientist.com>
23805 gst/playback/gstdecodebin.c: Handle the case where an element has multiple pads with unfixed caps as well as still po...
23806 Original commit message from CVS:
23807 * gst/playback/gstdecodebin.c: (find_dynamic), (dynamic_add),
23808 (close_pad_link), (elem_is_dynamic), (unlinked), (close_link):
23809 Handle the case where an element has multiple pads with
23810 unfixed caps as well as still possibly producing more dynamic
23811 pads by storing each case as a distinct entry in the dynamic list.
23812 Fixes #38223 again.
23814 2006-12-04 13:02:37 +0000 Wim Taymans <wim.taymans@gmail.com>
23816 gst/playback/gstdecodebin.c: Fix #382223, add more dynamic caps handling.
23817 Original commit message from CVS:
23818 * gst/playback/gstdecodebin.c: (close_pad_link):
23819 Fix #382223, add more dynamic caps handling.
23821 2006-12-01 11:35:57 +0000 Zaheer Abbas Merali <zaheerabbas@merali.org>
23824 Ignore all pot files
23825 Original commit message from CVS:
23826 Ignore all pot files
23828 2006-12-01 10:36:50 +0000 Michael Smith <msmith@xiph.org>
23830 gst/audiorate/gstaudiorate.c: Delete bad debug code.
23831 Original commit message from CVS:
23832 * gst/audiorate/gstaudiorate.c: (gst_audio_rate_chain):
23833 Delete bad debug code.
23836 2006-12-01 10:27:54 +0000 Sergey Scobich <sergey.scobich@gmail.com>
23838 Fix compilation on win32 under VS8
23839 Original commit message from CVS:
23840 * gst/videoscale/vs_4tap.c:
23842 * win32/common/config.h:
23843 * win32/vs8/libgstvideoscale.vcproj:
23844 Fix compilation on win32 under VS8
23845 Patch by: Sergey Scobich <sergey dot scobich at gmail dot com>
23846 Partially fixes #381175
23848 2006-11-30 23:46:27 +0000 Jan Schmidt <thaytan@mad.scientist.com>
23865 Original commit message from CVS:
23868 2006-11-30 12:50:42 +0000 Michael Smith <msmith@xiph.org>
23870 tests/check/pipelines/theoraenc.c: It would be very bad if, after a discont buffer, we thought every single following...
23871 Original commit message from CVS:
23872 * tests/check/pipelines/theoraenc.c: (check_buffer_granulepos),
23874 It would be very bad if, after a discont buffer, we thought every
23875 single following buffer was also discont. So, add to the test to
23876 ensure that this isn't the case.
23877 * ext/theora/theoraenc.c: (theora_enc_is_discontinuous):
23878 ... it was the case. So fix it.
23880 2006-11-28 16:43:18 +0000 Wim Taymans <wim.taymans@gmail.com>
23882 gst/playback/gstplaybasebin.c: Improve debug.
23883 Original commit message from CVS:
23884 * gst/playback/gstplaybasebin.c: (check_queue_event):
23886 * gst/videoscale/gstvideoscale.c: (gst_video_scale_transform_caps):
23887 Fix width and height range from 16 - 4096 to 1 - MAXINT, just like the
23888 padtemplate caps. Refixes #357577.
23890 2006-11-28 16:21:27 +0000 Wim Taymans <wim.taymans@gmail.com>
23892 gst/playback/gstplaybasebin.c: Add event probe to see when EOS is in a queue and we can disable the underrun signals....
23893 Original commit message from CVS:
23894 * gst/playback/gstplaybasebin.c: (check_queue_event),
23895 (queue_threshold_reached), (queue_out_of_data),
23896 (gen_preroll_element):
23897 Add event probe to see when EOS is in a queue and we can disable the
23898 underrun signals. Fixes #357577.
23900 2006-11-28 14:40:39 +0000 Edward Hervey <bilboed@bilboed.com>
23902 gst/playback/: New decodebin2 element.
23903 Original commit message from CVS:
23904 * gst/playback/Makefile.am:
23905 * gst/playback/gstdecodebin2.c: (gst_decode_bin_get_type),
23906 (_gst_boolean_accumulator), (gst_decode_bin_class_init),
23907 (gst_decode_bin_factory_filter), (compare_ranks), (print_feature),
23908 (gst_decode_bin_init), (gst_decode_bin_dispose),
23909 (gst_decode_bin_finalize), (gst_decode_bin_set_property),
23910 (gst_decode_bin_get_property), (gst_decode_bin_set_caps),
23911 (gst_decode_bin_get_caps), (gst_decode_bin_autoplug_continue),
23912 (gst_decode_bin_autoplug_sort), (analyze_new_pad), (connect_pad),
23913 (connect_element), (expose_pad), (type_found),
23914 (pad_added_group_cb), (pad_removed_group_cb),
23915 (no_more_pads_group_cb), (pad_added_cb), (pad_removed_cb),
23916 (no_more_pads_cb), (find_compatibles), (is_demuxer_element),
23917 (are_raw_caps), (multi_queue_overrun_cb),
23918 (multi_queue_underrun_cb), (gst_decode_group_new),
23919 (get_current_group), (group_demuxer_event_probe),
23920 (gst_decode_group_control_demuxer_pad),
23921 (gst_decode_group_control_source_pad),
23922 (gst_decode_group_check_if_blocked),
23923 (gst_decode_group_check_if_drained), (gst_decode_group_expose),
23924 (gst_decode_group_hide), (gst_decode_group_free),
23925 (gst_decode_group_set_complete), (source_pad_blocked_cb),
23926 (source_pad_event_probe), (gst_decode_pad_new), (add_fakesink),
23927 (remove_fakesink), (find_sink_pad), (gst_decode_bin_change_state),
23929 New decodebin2 element.
23931 * gst/playback/gstplay-marshal.list:
23932 Added marshallers for new signals in decodebin2
23933 * gst/playback/gstplaybasebin.c: (setup_subtitle), (make_decoder):
23934 Use decodebin2 if *and only if* the USE_DECODEBIN2 environment variable
23937 2006-11-28 10:45:40 +0000 Wim Taymans <wim.taymans@gmail.com>
23939 gst/playback/gstplaybasebin.c: Disable rtsp:// uris for the release, it's not good enough yet.
23940 Original commit message from CVS:
23941 * gst/playback/gstplaybasebin.c: (setup_source),
23942 (gst_play_base_bin_change_state):
23943 Disable rtsp:// uris for the release, it's not good enough yet.
23946 2006-11-26 16:39:41 +0000 Wim Taymans <wim.taymans@gmail.com>
23948 ext/theora/theoradec.c: Implement reverse playback.
23949 Original commit message from CVS:
23950 * ext/theora/theoradec.c: (gst_theora_dec_reset),
23951 (theora_dec_push_forward), (theora_dec_push_reverse),
23952 (theora_handle_data_packet), (theora_dec_decode_buffer),
23953 (theora_dec_flush_decode), (theora_dec_chain_reverse),
23954 (theora_dec_chain_forward), (theora_dec_chain):
23955 Implement reverse playback.
23956 * ext/vorbis/vorbisdec.c: (gst_vorbis_dec_reset),
23957 (vorbis_dec_decode_buffer), (vorbis_dec_flush_decode),
23958 (vorbis_dec_chain_forward):
23959 Clear buffers used for reverse playback in _reset.
23960 No need to set the eos flag, we clip samples using the segment.
23962 2006-11-24 15:40:58 +0000 Wim Taymans <wim.taymans@gmail.com>
23964 ext/ogg/gstoggdemux.c: Some cleanups.
23965 Original commit message from CVS:
23966 * ext/ogg/gstoggdemux.c: (gst_ogg_page_copy), (gst_ogg_page_free),
23967 (gst_ogg_pad_init), (gst_ogg_pad_dispose), (gst_ogg_pad_reset),
23968 (gst_ogg_pad_stream_out), (gst_ogg_pad_submit_page),
23969 (gst_ogg_chain_reset), (gst_ogg_demux_perform_seek):
23971 Handle continued pages in reverse mode.
23973 2006-11-24 15:39:03 +0000 Wim Taymans <wim.taymans@gmail.com>
23975 ext/vorbis/vorbisdec.c: Small cleanups.
23976 Original commit message from CVS:
23977 * ext/vorbis/vorbisdec.c: (vorbis_dec_push_forward),
23978 (vorbis_handle_data_packet), (vorbis_dec_decode_buffer),
23979 (vorbis_dec_flush_decode):
23981 Don't try to add invalid timestamps.
23982 Clipping will unref the buffer.
23984 2006-11-24 08:56:10 +0000 Stefan Kost <ensonic@users.sourceforge.net>
23986 gst/: remove obsolete _factory_init protos
23987 Original commit message from CVS:
23988 * gst/adder/gstadder.h:
23989 * gst/audiotestsrc/gstaudiotestsrc.h:
23990 remove obsolete _factory_init protos
23992 2006-11-24 08:35:23 +0000 Stefan Kost <ensonic@users.sourceforge.net>
23994 sys/xvimage/xvimagesink.c: Fix spacing in debug message.
23995 Original commit message from CVS:
23996 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_buffer_alloc):
23997 Fix spacing in debug message.
23999 2006-11-23 11:07:23 +0000 Wim Taymans <wim.taymans@gmail.com>
24001 ext/ogg/gstoggdemux.c: Don't just ignore return values from _pad_push().
24002 Original commit message from CVS:
24003 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_page),
24004 (gst_ogg_demux_chain):
24005 Don't just ignore return values from _pad_push().
24006 Small debug improvements.
24008 2006-11-23 11:02:11 +0000 Michael Smith <msmith@xiph.org>
24010 ext/ogg/gstoggmux.c: If our incoming buffer is marked as DISCONT, then increment the page number (so that the discont...
24011 Original commit message from CVS:
24012 * ext/ogg/gstoggmux.c: (gst_ogg_mux_process_best_pad):
24013 If our incoming buffer is marked as DISCONT, then increment the page
24014 number (so that the discontinuity is marked in the final ogg
24015 bitstream) and flush the previous page.
24017 2006-11-22 14:34:03 +0000 Michael Smith <msmith@xiph.org>
24019 ext/theora/: Mark discontinuities of > 3/4 of a frame, reinit encoder.
24020 Original commit message from CVS:
24021 * ext/theora/gsttheoraenc.h:
24022 * ext/theora/theoraenc.c: (gst_theora_enc_init),
24023 (theora_enc_reset), (theora_enc_clear), (theora_enc_sink_setcaps),
24024 (theora_buffer_from_packet), (theora_enc_is_discontinuous),
24025 (theora_enc_chain), (theora_enc_change_state):
24026 Mark discontinuities of > 3/4 of a frame, reinit encoder.
24027 * tests/check/pipelines/theoraenc.c: (check_buffer_granulepos),
24028 (GST_START_TEST), (theoraenc_suite):
24029 Enable discontinuity test, fix it.
24031 2006-11-21 18:39:34 +0000 Tim-Philipp Müller <tim@centricular.net>
24033 ext/pango/gsttextoverlay.*: Some textoverlay fixes: for one, in the video chain function, actually wait for a text bu...
24034 Original commit message from CVS:
24035 * ext/pango/gsttextoverlay.c: (gst_text_overlay_init),
24036 (gst_text_overlay_text_pad_unlink), (gst_text_overlay_text_event),
24037 (gst_text_overlay_video_event), (gst_text_overlay_pop_text),
24038 (gst_text_overlay_text_chain), (gst_text_overlay_video_chain),
24039 (gst_text_overlay_change_state):
24040 * ext/pango/gsttextoverlay.h:
24041 Some textoverlay fixes: for one, in the video chain function,
24042 actually wait for a text buffer to come in if there is none at the
24043 moment and there should be one; also, deal more gracefully with
24044 incoming buffers that do not have a timestamp or duration; discard
24045 text buffer when not needed any longer. Fixes #341681.
24046 * tests/check/Makefile.am:
24047 * tests/check/elements/.cvsignore:
24048 * tests/check/elements/textoverlay.c:
24049 (notgst_check_setup_src_pad2), (notgst_check_teardown_src_pad2),
24050 (setup_textoverlay), (buffer_is_all_black), (create_black_buffer),
24051 (create_text_buffer), (cleanup_textoverlay), (GST_START_TEST),
24052 (test_video_waits_for_text_send_text_newsegment_thread),
24053 (test_video_waits_for_text_shutdown_element),
24054 (test_render_continuity_push_video_buffers_thread),
24055 (textoverlay_suite):
24056 Add some unit tests for textoverlay.
24058 2006-11-21 09:29:56 +0000 Tim-Philipp Müller <tim@centricular.net>
24060 gst/typefind/gsttypefindfunctions.c: Avoid integer underflow when the found probability for mp3 is smaller than the '...
24061 Original commit message from CVS:
24062 * gst/typefind/gsttypefindfunctions.c: (mp3_type_find_at_offset):
24063 Avoid integer underflow when the found probability for mp3 is
24064 smaller than the 'penalty' we subtract if there's not a clean
24065 mp3 header sync at offset 0.
24067 2006-11-21 08:17:16 +0000 Stefan Kost <ensonic@users.sourceforge.net>
24069 docs/libs/gst-plugins-base-libs-sections.txt: Add some new symbols to the docs
24070 Original commit message from CVS:
24071 * docs/libs/gst-plugins-base-libs-sections.txt:
24072 Add some new symbols to the docs
24074 2006-11-20 16:44:28 +0000 Tim-Philipp Müller <tim@centricular.net>
24076 tests/check/: Enable ffmpegcolorspace test now that the RGBA32 issue is fixed (for now not for valgrinding though, si...
24077 Original commit message from CVS:
24078 * tests/check/Makefile.am:
24079 * tests/check/elements/ffmpegcolorspace.c:
24080 (ffmpegcolorspace_suite):
24081 Enable ffmpegcolorspace test now that the RGBA32 issue is fixed
24082 (for now not for valgrinding though, since it takes too long).
24084 2006-11-20 15:01:09 +0000 Wim Taymans <wim.taymans@gmail.com>
24086 gst/ffmpegcolorspace/gstffmpegcodecmap.c: Fix RGBA32 caps. Fixes #357038.
24087 Original commit message from CVS:
24088 * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
24089 (gst_ffmpeg_pixfmt_to_caps):
24090 Fix RGBA32 caps. Fixes #357038.
24092 2006-11-20 12:20:39 +0000 Tim-Philipp Müller <tim@centricular.net>
24094 gst-libs/gst/interfaces/mixertrack.h: Add FIXME so we can add some padding here in 0.11
24095 Original commit message from CVS:
24096 * gst-libs/gst/interfaces/mixertrack.h:
24097 Add FIXME so we can add some padding here in 0.11
24099 2006-11-19 17:07:34 +0000 Tim-Philipp Müller <tim@centricular.net>
24101 gst-libs/gst/rtp/gstbasertpaudiopayload.h: Fix GstBaseRTPAudioPayload structure so the whole GObject inheritance busi...
24102 Original commit message from CVS:
24103 * gst-libs/gst/rtp/gstbasertpaudiopayload.h:
24104 Fix GstBaseRTPAudioPayload structure so the whole GObject
24105 inheritance business actually works (parent class instance structure
24106 must always come first in the derived class instance structure).
24108 2006-11-16 14:35:30 +0000 Tim-Philipp Müller <tim@centricular.net>
24110 Make sure our checks and the videotestsrc plugin link against the local uninstalled gst libs and not any installed gs...
24111 Original commit message from CVS:
24112 * gst/videotestsrc/Makefile.am:
24113 * tests/check/Makefile.am:
24114 Make sure our checks and the videotestsrc plugin link against the
24115 local uninstalled gst libs and not any installed gst libs that
24116 might happen to exist as well.
24117 * tests/check/elements/adder.c: (message_received),
24118 (test_event_message_received), (test_play_twice_message_received):
24119 * tests/check/elements/ffmpegcolorspace.c: (GST_START_TEST):
24120 Fix compiler warnings when compiling against core with disabled
24123 2006-11-16 12:55:08 +0000 Michael Smith <msmith@xiph.org>
24125 gst/audiorate/gstaudiorate.c: Fix audiorate, so that it accurately sets offsets and timestamps.
24126 Original commit message from CVS:
24127 * gst/audiorate/gstaudiorate.c: (gst_audio_rate_reset),
24128 (gst_audio_rate_sink_event), (gst_audio_rate_chain):
24129 Fix audiorate, so that it accurately sets offsets and timestamps.
24130 Doesn't change the fundamental algorithmic decisions; so should be
24132 * tests/check/Makefile.am:
24133 Enable audiorate test now that it passes.
24135 2006-11-15 10:05:33 +0000 Stefan Kost <ensonic@users.sourceforge.net>
24137 sys/xvimage/xvimagesink.c: clear xv when going to NULL, remove // commented non-existant proto
24138 Original commit message from CVS:
24139 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_change_state):
24140 clear xv when going to NULL, remove // commented non-existant proto
24141 * tests/examples/seek/seek.c: (main):
24142 add missing tooltip description for scrub and play_scrub
24144 2006-11-14 23:34:19 +0000 David Schleef <ds@schleef.org>
24146 configure.ac: Bump liboil requirement to 0.3.8.
24147 Original commit message from CVS:
24149 Bump liboil requirement to 0.3.8.
24150 * gst-libs/gst/riff/riff-media.c:
24152 * gst/videoscale/vs_image.h:
24153 * gst/videoscale/vs_scanline.h:
24154 Use liboil's stdint.h.
24155 * gst/videotestsrc/videotestsrc.c:
24156 Remove liboil related ifdef's, since they aren't needed now, and
24157 won't work with future versions.
24159 2006-11-14 23:08:38 +0000 David Schleef <ds@schleef.org>
24161 gst/videoscale/: Add a 4-tap image scaler. Theoretically looks much prettier.
24162 Original commit message from CVS:
24163 * gst/videoscale/Makefile.am:
24164 * gst/videoscale/gstvideoscale.c:
24165 * gst/videoscale/gstvideoscale.h:
24166 * gst/videoscale/vs_4tap.c:
24167 * gst/videoscale/vs_4tap.h:
24168 * gst/videoscale/vs_image.c:
24169 * gst/videoscale/vs_image.h:
24170 * gst/videoscale/vs_scanline.c:
24171 * gst/videoscale/vs_scanline.h:
24172 Add a 4-tap image scaler. Theoretically looks much prettier.
24173 The tap calculation could use some improvement.
24175 2006-11-14 11:54:14 +0000 Jan David Mol <j.j.d.mol@tudelft.nl>
24177 Various gsize and gssize printf fixes. Fixes #372507.
24178 Original commit message from CVS:
24179 Patch by: Jan David Mol <j dot j dot d dot mol at tudelft dot nl>
24180 * gst-libs/gst/riff/riff-read.c: (gst_riff_parse_strf_auds),
24181 (gst_riff_parse_strf_iavs):
24182 * gst/subparse/gstsubparse.c: (convert_encoding):
24183 * gst/tcp/gstmultifdsink.c:
24184 (gst_multi_fd_sink_handle_client_write):
24185 * gst/tcp/gsttcp.c: (gst_tcp_socket_write), (gst_tcp_socket_read),
24186 (gst_tcp_read_buffer), (gst_tcp_gdp_read_caps),
24187 (gst_tcp_gdp_write_buffer), (gst_tcp_gdp_write_caps):
24188 * gst/tcp/gsttcpclientsink.c: (gst_tcp_client_sink_render):
24189 * sys/ximage/ximagesink.c: (gst_ximagesink_check_xshm_calls),
24190 (gst_ximagesink_ximage_new):
24191 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_new):
24192 Various gsize and gssize printf fixes. Fixes #372507.
24194 2006-11-13 18:14:48 +0000 Wim Taymans <wim.taymans@gmail.com>
24196 ext/vorbis/vorbisdec.*: First stab at vorbis reverse playback.
24197 Original commit message from CVS:
24198 * ext/vorbis/vorbisdec.c: (vorbis_dec_sink_event),
24199 (vorbis_dec_push_forward), (vorbis_dec_push_reverse),
24200 (vorbis_handle_data_packet), (vorbis_dec_decode_buffer),
24201 (vorbis_dec_flush_decode), (vorbis_dec_chain_reverse),
24202 (vorbis_dec_chain_forward), (vorbis_dec_chain):
24203 * ext/vorbis/vorbisdec.h:
24204 First stab at vorbis reverse playback.
24206 2006-11-13 17:30:17 +0000 Wim Taymans <wim.taymans@gmail.com>
24208 gst-libs/gst/audio/gstbaseaudiosink.*: Make the clock sync code more accurate wrt resampling and playback at differen...
24209 Original commit message from CVS:
24210 * gst-libs/gst/audio/gstbaseaudiosink.c:
24211 (gst_base_audio_sink_event), (gst_base_audio_sink_render):
24212 * gst-libs/gst/audio/gstbaseaudiosink.h:
24213 Make the clock sync code more accurate wrt resampling and playback
24214 at different rates.
24215 * gst-libs/gst/audio/gstringbuffer.c:
24216 (gst_ring_buffer_commit_full), (gst_ring_buffer_commit):
24217 * gst-libs/gst/audio/gstringbuffer.h:
24218 Use better algorithm to interpolate sample rates.
24220 2006-11-13 15:31:01 +0000 Michael Smith <msmith@xiph.org>
24222 ext/ogg/gstoggdemux.c: Improve a debug line slightly.
24223 Original commit message from CVS:
24224 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_page):
24225 Improve a debug line slightly.
24226 * ext/ogg/gstogmparse.c: (gst_ogm_parse_plugin_init):
24227 Call gst_riff_init() in plugin_init, to avoid getting errors from
24228 the debug system (unrelated changes to another plugin made this turn
24231 2006-11-10 19:20:21 +0000 Sergey Scobich <sergery.scobich@gmail.com>
24233 win32/common/libgsttag.def: Add missing symbol (#366492).
24234 Original commit message from CVS:
24235 Patch by: Sergey Scobich <sergery.scobich at gmail com>
24236 * win32/common/libgsttag.def:
24237 Add missing symbol (#366492).
24239 2006-11-10 00:52:55 +0000 Tim-Philipp Müller <tim@centricular.net>
24241 gst/playback/gststreamselector.c: Don't unref a NULL pad.
24242 Original commit message from CVS:
24243 * gst/playback/gststreamselector.c: (gst_stream_selector_dispose):
24244 Don't unref a NULL pad.
24246 2006-11-09 00:50:00 +0000 Christian Schaller <uraeus@gnome.org>
24248 ext/ogg/gstoggdemux.c: Implement first stab at reverse playback.
24249 Original commit message from CVS:
24250 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_page),
24251 (gst_ogg_demux_get_prev_page), (gst_ogg_demux_perform_seek),
24252 (gst_ogg_demux_handle_page), (gst_ogg_demux_chain),
24253 (gst_ogg_demux_loop_forward), (gst_ogg_demux_loop_reverse),
24254 (gst_ogg_demux_loop):
24255 Implement first stab at reverse playback.
24257 2006-11-07 07:22:32 +0000 Stefan Kost <ensonic@users.sourceforge.net>
24259 gst-libs/gst/riff/riff-media.c: add h263/h264 variants to the caps, Fixes #363118
24260 Original commit message from CVS:
24261 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
24262 (gst_riff_create_video_template_caps):
24263 add h263/h264 variants to the caps, Fixes #363118
24265 2006-11-06 18:24:59 +0000 Tim-Philipp Müller <tim@centricular.net>
24267 gst-libs/gst/audio/: Use g_strerror instead of strerror so we get UTF-8.
24268 Original commit message from CVS:
24269 * gst-libs/gst/audio/gstaudiosink.c: (audioringbuffer_thread_func):
24270 * gst-libs/gst/audio/gstaudiosrc.c: (audioringbuffer_thread_func):
24271 Use g_strerror instead of strerror so we get UTF-8.
24273 2006-11-04 07:25:58 +0000 David Schleef <ds@schleef.org>
24275 ext/ogg/: Add/remove KW-DIRAC header here, since it is ogg-specific.
24276 Original commit message from CVS:
24277 * ext/ogg/gstoggdemux.c:
24278 * ext/ogg/gstoggmux.c:
24279 Add/remove KW-DIRAC header here, since it is ogg-specific.
24281 2006-11-03 15:44:31 +0000 Michael Smith <msmith@xiph.org>
24283 gst/typefind/gsttypefindfunctions.c: Recognise more mpeg4 elementary video streams.
24284 Original commit message from CVS:
24285 * gst/typefind/gsttypefindfunctions.c: (mpeg4_video_type_find):
24286 Recognise more mpeg4 elementary video streams.
24288 2006-11-02 17:26:03 +0000 Edward Hervey <bilboed@bilboed.com>
24290 gst/typefind/gsttypefindfunctions.c: Lower the probability of mp3 typefinding functions if we don't find a valid mp3 ...
24291 Original commit message from CVS:
24292 * gst/typefind/gsttypefindfunctions.c: (mp3_type_find_at_offset):
24293 Lower the probability of mp3 typefinding functions if we don't find a
24294 valid mp3 header at the start of the file.
24297 2006-11-02 15:06:36 +0000 Wim Taymans <wim.taymans@gmail.com>
24299 ext/theora/: Document and partially implement an algorithm for doing reverse playback of theora video.
24300 Original commit message from CVS:
24301 * ext/theora/gsttheoradec.h:
24302 * ext/theora/theoradec.c: (gst_theora_dec_init),
24303 (theora_dec_sink_event), (theora_dec_chain_forward),
24304 (theora_dec_flush_decode), (theora_dec_chain_reverse),
24305 (theora_dec_chain):
24306 Document and partially implement an algorithm for doing reverse playback
24309 2006-11-02 14:18:45 +0000 Sergey Scobich <sergey.scobich@gmail.com>
24311 win32/: Misc. VS8 build fixes: fix syntax in config.h, add missing entries to libgsttag.def; add missing dependencies...
24312 Original commit message from CVS:
24313 Patch by: Sergey Scobich <sergey.scobich at gmail com>
24314 * win32/common/config.h:
24315 * win32/common/interfaces-enumtypes.c:
24316 * win32/common/libgsttag.def:
24317 * win32/vs8/gst-plugins-base.sln:
24318 * win32/vs8/libgstaudioresample.vcproj:
24319 * win32/vs8/libgstinterfaces.vcproj:
24320 * win32/vs8/libgstogg.vcproj:
24321 * win32/vs8/libgstriff.vcproj:
24322 * win32/vs8/libgsttag.vcproj:
24323 * win32/vs8/libgsttheora.vcproj:
24324 * win32/vs8/libgstvideoscale.vcproj:
24325 * win32/vs8/libgstvorbis.vcproj:
24326 Misc. VS8 build fixes: fix syntax in config.h, add missing entries
24327 to libgsttag.def; add missing dependencies for some vs8 projects;
24328 re-arrange placement of .def files in vs8 projects (#366334).
24330 2006-11-01 14:08:31 +0000 Tim-Philipp Müller <tim@centricular.net>
24332 ext/ogg/gstogg.c: Remove unused variable.
24333 Original commit message from CVS:
24334 * ext/ogg/gstogg.c:
24335 Remove unused variable.
24336 * ext/ogg/gstoggdemux.c:
24337 Fix Wim's surname in plugin description.
24339 2006-10-31 15:05:33 +0000 Wim Taymans <wim.taymans@gmail.com>
24341 gst-plugins-base.spec.in: spec new .h file. Fixes #368310.
24342 Original commit message from CVS:
24343 * gst-plugins-base.spec.in:
24344 spec new .h file. Fixes #368310.
24346 2006-10-31 14:19:07 +0000 Michael Smith <msmith@xiph.org>
24348 gst/tcp/gstmultifdsink.*: Make using the remove or clear signals threadsafe.
24349 Original commit message from CVS:
24350 * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_add_full),
24351 (gst_multi_fd_sink_remove), (gst_multi_fd_sink_clear),
24352 (gst_multi_fd_sink_get_stats),
24353 (gst_multi_fd_sink_remove_client_link),
24354 (gst_multi_fd_sink_queue_buffer),
24355 (gst_multi_fd_sink_handle_clients):
24356 * gst/tcp/gstmultifdsink.h:
24357 Make using the remove or clear signals threadsafe.
24358 Make calling get-stats with an invalid fd not segfault.
24361 2006-10-31 10:49:19 +0000 Wim Taymans <wim.taymans@gmail.com>
24363 gst-libs/gst/rtp/: Fix and activate base audio payloader.
24364 Original commit message from CVS:
24365 * gst-libs/gst/rtp/Makefile.am:
24366 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
24367 (gst_base_rtp_audio_payload_init):
24368 Fix and activate base audio payloader.
24370 2006-10-28 17:22:57 +0000 Tim-Philipp Müller <tim@centricular.net>
24372 gst/typefind/gsttypefindfunctions.c: Add typefinder for QuickTime Image Files (see #366156).
24373 Original commit message from CVS:
24374 * gst/typefind/gsttypefindfunctions.c: (qtif_type_find),
24376 Add typefinder for QuickTime Image Files (see #366156).
24378 2006-10-28 16:00:51 +0000 Tim-Philipp Müller <tim@centricular.net>
24380 gst/audioresample/gstaudioresample.c: Another typo fix (#366212).
24381 Original commit message from CVS:
24382 * gst/audioresample/gstaudioresample.c: (gst_audioresample_init):
24383 Another typo fix (#366212).
24385 2006-10-27 17:13:35 +0000 Wim Taymans <wim.taymans@gmail.com>
24387 gst/volume/gstvolume.c: Use stream time to synchronize volume property instead of rather random timestamps. This is n...
24388 Original commit message from CVS:
24389 * gst/volume/gstvolume.c: (volume_transform_ip):
24390 Use stream time to synchronize volume property instead of rather random
24391 timestamps. This is needed when gnonlin does its time shifting.
24393 2006-10-27 16:46:15 +0000 Wim Taymans <wim.taymans@gmail.com>
24396 I'm too lazy to comment this
24397 Original commit message from CVS:
24398 *** empty log message ***
24400 2006-10-27 16:45:30 +0000 Mark Nauwelaerts <manauw@skynet.be>
24402 ext/ogg/gstoggmux.c: Remove the pad from the element in release_pad.
24403 Original commit message from CVS:
24404 Patch by: Mark Nauwelaerts <manauw at skynet dot be>
24405 * ext/ogg/gstoggmux.c: (gst_ogg_mux_release_pad):
24406 Remove the pad from the element in release_pad.
24408 2006-10-27 11:57:18 +0000 Tim-Philipp Müller <tim@centricular.net>
24410 sys/: Explicitly create our custom buffer classes at a thread-safe location as well, since g_type_class_ref() doesn't...
24411 Original commit message from CVS:
24412 * sys/ximage/ximagesink.c: (gst_ximagesink_get_type):
24413 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_type):
24414 Explicitly create our custom buffer classes at a thread-safe
24415 location as well, since g_type_class_ref() doesn't seem to be
24416 entirely thread-safe either (#365501; also see #349410).
24418 2006-10-26 10:49:00 +0000 Tim-Philipp Müller <tim@centricular.net>
24420 gst-libs/gst/riff/riff-read.c: If strings in INFO chunk are not UTF-8, do something similar to what we do for ID3v1 t...
24421 Original commit message from CVS:
24422 * gst-libs/gst/riff/riff-read.c: (freeform_string_to_utf8),
24423 (gst_riff_parse_info):
24424 If strings in INFO chunk are not UTF-8, do something similar to
24425 what we do for ID3v1 tags: check a number of environment variables
24426 (GST_AVI_TAG_ENCODING, GST_RIFF_TAG_ENCODING, GST_TAG_ENCODING) for
24427 character sets to try, otherwise try the current locale and/or fall
24428 back on ISO-8859-1. Fixes #360552.
24430 2006-10-23 12:46:41 +0000 Tim-Philipp Müller <tim@centricular.net>
24432 gst/videotestsrc/: Add a bunch of exciting new checkers patterns.
24433 Original commit message from CVS:
24434 * gst/videotestsrc/gstvideotestsrc.c:
24435 (gst_video_test_src_pattern_get_type),
24436 (gst_video_test_src_set_pattern):
24437 * gst/videotestsrc/gstvideotestsrc.h:
24438 * gst/videotestsrc/videotestsrc.c: (gst_video_test_src_checkers1),
24439 (gst_video_test_src_checkers2), (gst_video_test_src_checkers4),
24440 (gst_video_test_src_checkers8):
24441 * gst/videotestsrc/videotestsrc.h:
24442 Add a bunch of exciting new checkers patterns.
24444 2006-10-23 12:06:44 +0000 Tim-Philipp Müller <tim@centricular.net>
24446 gst/subparse/: Add support for TMPlayer-type subtitles (#362845).
24447 Original commit message from CVS:
24448 * gst/subparse/Makefile.am:
24449 * gst/subparse/gstsubparse.c:
24450 (gst_sub_parse_data_format_autodetect),
24451 (gst_sub_parse_format_autodetect), (handle_buffer),
24452 (gst_sub_parse_chain), (gst_subparse_type_find), (plugin_init):
24453 * gst/subparse/gstsubparse.h:
24454 * gst/subparse/tmplayerparse.c: (tmplayer_parse_line),
24456 * gst/subparse/tmplayerparse.h:
24457 Add support for TMPlayer-type subtitles (#362845).
24458 * tests/check/elements/subparse.c: (test_tmplayer_do_test),
24459 (GST_START_TEST), (subparse_suite):
24460 Add some basic unit tests for the above.
24462 2006-10-23 11:47:04 +0000 Tim-Philipp Müller <tim@centricular.net>
24464 tests/check/elements/audiorate.c: More tests for audiorate: inject buffers to check behaviour when buffers overlap.
24465 Original commit message from CVS:
24466 * tests/check/elements/audiorate.c: (test_injector_base_init),
24467 (test_injector_class_init), (test_injector_chain),
24468 (test_injector_init), (probe_cb), (do_perfect_stream_test),
24469 (GST_START_TEST), (audiorate_suite):
24470 More tests for audiorate: inject buffers to check behaviour when
24473 2006-10-21 16:39:54 +0000 Tim-Philipp Müller <tim@centricular.net>
24475 tests/check/: Add some basic unit tests for audiorate. Disabled at the moment since it doesn't pass yet (see bug #363...
24476 Original commit message from CVS:
24477 * tests/check/Makefile.am:
24478 * tests/check/elements/.cvsignore:
24479 * tests/check/elements/audiorate.c: (probe_cb), (got_buf),
24480 (do_perfect_stream_test), (GST_START_TEST), (audiorate_suite):
24481 Add some basic unit tests for audiorate. Disabled at the moment
24482 since it doesn't pass yet (see bug #363119).
24484 2006-10-20 17:02:19 +0000 Tim-Philipp Müller <tim@centricular.net>
24486 gst/subparse/gstsubparse.c: Add missing closing tags for markup and fix broken markup, otherwise pango won't render a...
24487 Original commit message from CVS:
24488 * gst/subparse/gstsubparse.c: (subrip_fix_up_markup),
24489 (parse_subrip), (handle_buffer):
24490 Add missing closing tags for markup and fix broken markup,
24491 otherwise pango won't render anything (fixes #357531). Also,
24492 make sure the text we send out is always NUL-terminated
24493 (better safe than sorry etc.).
24494 * tests/check/elements/subparse.c: (test_srt_do_test),
24496 Some more tests for .srt incl. tests for the above stuff.
24498 2006-10-20 13:56:55 +0000 Stefan Kost <ensonic@users.sf.net>
24500 sys/: Try to redraw borders only when needed. Apparently this consumes resources on small devices... :-O (#363607)
24501 Original commit message from CVS:
24502 2006-10-20 Julien MOUTTE <julien@moutte.net>
24503 * sys/ximage/ximagesink.c: (gst_ximagesink_ximage_put):
24504 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_put):
24505 Patch by: Stefan Kost <ensonic@users.sf.net>
24506 Try to redraw borders only when needed. Apparently this consumes
24507 resources on small devices... :-O (#363607)
24509 2006-10-20 13:54:19 +0000 Michael Smith <msmith@xiph.org>
24511 gst/tcp/gstmultifdsink.c: If caps change, then update the client's idea of the caps so that we don't end up re-sendin...
24512 Original commit message from CVS:
24513 * gst/tcp/gstmultifdsink.c:
24514 (gst_multi_fd_sink_client_queue_buffer):
24515 If caps change, then update the client's idea of the caps so that we
24516 don't end up re-sending streamheaders for every single buffer after
24519 2006-10-20 12:31:02 +0000 Michael Smith <msmith@xiph.org>
24521 ext/ogg/gstoggparse.c: Set caps on pushed buffers; fix up refcounting of caps objects.
24522 Original commit message from CVS:
24523 * ext/ogg/gstoggparse.c: (gst_ogg_parse_dispose),
24524 (gst_ogg_parse_append_header), (gst_ogg_parse_chain):
24525 Set caps on pushed buffers; fix up refcounting of caps objects.
24527 2006-10-19 14:09:30 +0000 Tim-Philipp Müller <tim@centricular.net>
24529 gst/typefind/gsttypefindfunctions.c: Typefind mmsh header data packet to application/x-mmsh (#362625).
24530 Original commit message from CVS:
24531 * gst/typefind/gsttypefindfunctions.c: (mmsh_type_find),
24533 Typefind mmsh header data packet to application/x-mmsh (#362625).
24535 2006-10-19 09:17:48 +0000 Tim-Philipp Müller <tim@centricular.net>
24537 tests/check/: Add very simple unit test for subparse.
24538 Original commit message from CVS:
24539 * tests/check/Makefile.am:
24540 * tests/check/elements/.cvsignore:
24541 * tests/check/elements/subparse.c: (buffer_from_static_string),
24542 (setup_subparse), (teardown_subparse), (test_srt_do_test),
24543 (GST_START_TEST), (subparse_suite):
24544 Add very simple unit test for subparse.
24546 2006-10-19 09:00:21 +0000 Tim-Philipp Müller <tim@centricular.net>
24548 gst/subparse/gstsubparse.c: Strip trailing newlines from subtitle text output.
24549 Original commit message from CVS:
24550 * gst/subparse/gstsubparse.c: (strip_trailing_newlines),
24552 Strip trailing newlines from subtitle text output.
24554 2006-10-18 18:40:12 +0000 Tim-Philipp Müller <tim@centricular.net>
24556 gst/subparse/gstsubparse.c: Fix memleak; clear subparse->textbuf n state change function.
24557 Original commit message from CVS:
24558 * gst/subparse/gstsubparse.c: (gst_sub_parse_dispose),
24559 (gst_sub_parse_change_state):
24560 Fix memleak; clear subparse->textbuf n state change function.
24562 2006-10-18 15:13:59 +0000 Tim-Philipp Müller <tim@centricular.net>
24564 gst/subparse/gstsubparse.c: Don't require subrip (.srt) files to start with a chunk number of 1.
24565 Original commit message from CVS:
24566 * gst/subparse/gstsubparse.c:
24567 (gst_sub_parse_data_format_autodetect):
24568 Don't require subrip (.srt) files to start with a chunk number of 1.
24570 2006-10-18 13:42:49 +0000 Wim Taymans <wim.taymans@gmail.com>
24572 gst-libs/gst/audio/gstbaseaudiosink.*: Extract rate from the NEWSEGMENT event.
24573 Original commit message from CVS:
24574 * gst-libs/gst/audio/gstbaseaudiosink.c:
24575 (gst_base_audio_sink_event), (gst_base_audio_sink_render):
24576 * gst-libs/gst/audio/gstbaseaudiosink.h:
24577 Extract rate from the NEWSEGMENT event.
24578 Use commit_full to also take rate adjustment into account when writing
24579 samples to the ringbuffer.
24580 * gst-libs/gst/audio/gstringbuffer.c:
24581 (gst_ring_buffer_commit_full), (gst_ring_buffer_commit),
24582 (gst_ring_buffer_read):
24583 * gst-libs/gst/audio/gstringbuffer.h:
24584 Added _commit_full() to also take rate into account.
24585 Use simple interpolation algorithm to resample audio.
24586 API: gst_ring_buffer_commit_full()
24587 * tests/examples/seek/scrubby.c: (speed_cb), (do_seek):
24588 * tests/examples/seek/seek.c: (segment_done):
24589 Don't try to seek with 0.0 rate, just pause instead.
24590 Remove bogus debug line.
24592 2006-10-18 12:57:54 +0000 Tim-Philipp Müller <tim@centricular.net>
24594 gst/playback/gstplaybasebin.c: Catch async errors when starting up the subtitle bin, so we can stop waiting and conti...
24595 Original commit message from CVS:
24596 * gst/playback/gstplaybasebin.c: (subbin_startup_sync_msg),
24598 Catch async errors when starting up the subtitle bin, so we can
24599 stop waiting and continue with the main film instead of hanging
24600 forever. Fixes #339366.
24601 * tests/check/elements/playbin.c: (playbin_suite):
24602 Enable unit test for the above.
24604 2006-10-18 09:53:03 +0000 Tim-Philipp Müller <tim@centricular.net>
24606 tests/check/: Some small and basic unit tests for playbin; not very useful yet, but at least a start.
24607 Original commit message from CVS:
24608 * tests/check/Makefile.am:
24609 * tests/check/elements/.cvsignore:
24610 * tests/check/elements/playbin.c: (GST_START_TEST),
24611 (gst_red_video_src_uri_get_type),
24612 (gst_red_video_src_uri_get_protocols),
24613 (gst_red_video_src_uri_get_uri), (gst_red_video_src_uri_set_uri),
24614 (gst_red_video_src_uri_handler_init),
24615 (gst_red_video_src_init_type), (gst_red_video_src_base_init),
24616 (gst_red_video_src_create), (gst_red_video_src_class_init),
24617 (gst_red_video_src_init), (plugin_init), (playbin_suite):
24618 Some small and basic unit tests for playbin; not very useful yet,
24619 but at least a start.
24621 2006-10-18 09:46:35 +0000 Tim-Philipp Müller <tim@centricular.net>
24623 gst/playback/gstplaybin.c: The old pad activation spiel.
24624 Original commit message from CVS:
24625 * gst/playback/gstplaybin.c: (setup_sinks):
24626 The old pad activation spiel.
24628 2006-10-18 09:31:49 +0000 Tim-Philipp Müller <tim@centricular.net>
24630 gst/playback/gstplaybasebin.c: Don't hang forever if the subbin already fails to start up in the state change to PAUS...
24631 Original commit message from CVS:
24632 * gst/playback/gstplaybasebin.c: (setup_source):
24633 Don't hang forever if the subbin already fails to start up in
24634 the state change to PAUSED (#339366).
24636 2006-10-17 17:17:16 +0000 Tim-Philipp Müller <tim@centricular.net>
24638 gst-libs/gst/interfaces/tuner.c: Fix some function guards, add some more function guards.
24639 Original commit message from CVS:
24640 * gst-libs/gst/interfaces/tuner.c: (gst_tuner_list_channels),
24641 (gst_tuner_set_channel), (gst_tuner_get_channel),
24642 (gst_tuner_list_norms), (gst_tuner_set_norm), (gst_tuner_get_norm),
24643 (gst_tuner_set_frequency), (gst_tuner_get_frequency),
24644 (gst_tuner_signal_strength), (gst_tuner_find_norm_by_name),
24645 (gst_tuner_find_channel_by_name):
24646 Fix some function guards, add some more function guards.
24648 2006-10-17 11:34:32 +0000 Jan Schmidt <thaytan@mad.scientist.com>
24650 gst/playback/gstdecodebin.c: Don't return a pad from get_our_ghost_pad unless it is actually the one we want.
24651 Original commit message from CVS:
24652 * gst/playback/gstdecodebin.c: (get_our_ghost_pad),
24653 (remove_element_chain):
24654 Don't return a pad from get_our_ghost_pad unless it is actually the
24656 Change a cast in remove_element_chain slightly.
24658 2006-10-13 15:20:29 +0000 Julien Moutte <julien@moutte.net>
24660 tests/examples/seek/seek.c: Segment seeking needs to use the rate and set stop to -1.
24661 Original commit message from CVS:
24662 2006-10-13 Julien MOUTTE <julien@moutte.net>
24663 * tests/examples/seek/seek.c: (do_seek), (start_seek),
24664 (rate_spinbutton_changed_cb), (segment_done),
24665 (msg_state_changed):
24666 Segment seeking needs to use the rate and set stop to -1.
24668 2006-10-13 14:15:42 +0000 Ville Syrjala <ville.syrjala@movial.fi>
24670 gst-libs/gst/audio/gstbaseaudiosink.c: Don't crash when ringbuffer is not yet created.
24671 Original commit message from CVS:
24672 * gst-libs/gst/audio/gstbaseaudiosink.c:
24673 (gst_base_audio_sink_setcaps):
24674 Don't crash when ringbuffer is not yet created.
24675 Patch by: Ville Syrjala <ville dot syrjala at movial dot fi>
24677 * gst/playback/gstplaybasebin.c: (new_decoded_pad_full):
24678 * gst/playback/gststreamselector.c:
24679 (gst_stream_selector_request_new_pad):
24680 Activate pads befre adding them to running elements.
24682 2006-10-13 11:25:10 +0000 Julien Moutte <julien@moutte.net>
24684 tests/examples/seek/seek.c: Stop the scale updater when we start grabing the slider. Don't wait for the pipeline to b...
24685 Original commit message from CVS:
24686 2006-10-13 Julien MOUTTE <julien@moutte.net>
24687 * tests/examples/seek/seek.c: (do_seek), (start_seek),
24688 (rate_spinbutton_changed_cb), (msg_state_changed): Stop the
24690 updater when we start grabing the slider. Don't wait for the
24691 pipeline to be PAUSED.
24693 2006-10-13 08:57:52 +0000 Tim-Philipp Müller <tim@centricular.net>
24695 gst-libs/gst/interfaces/mixer.c: Guard mixer interface functions against bogus arguments.
24696 Original commit message from CVS:
24697 * gst-libs/gst/interfaces/mixer.c: (gst_mixer_list_tracks),
24698 (gst_mixer_set_volume), (gst_mixer_get_volume),
24699 (gst_mixer_set_mute), (gst_mixer_set_option),
24700 (gst_mixer_get_option), (gst_mixer_mute_toggled),
24701 (gst_mixer_record_toggled), (gst_mixer_volume_changed),
24702 (gst_mixer_option_changed):
24703 Guard mixer interface functions against bogus arguments.
24705 2006-10-12 19:39:07 +0000 Julien Moutte <julien@moutte.net>
24707 tests/examples/seek/seek.c: Use state-changed messages to trigger start/stop of scale update timer. Indeed the scale ...
24708 Original commit message from CVS:
24709 2006-10-12 Julien MOUTTE <julien@moutte.net>
24710 * tests/examples/seek/seek.c: (do_seek), (start_seek),
24712 (play_cb), (pause_cb), (stop_cb),
24713 (rate_spinbutton_changed_cb),
24714 (msg_state_changed), (main): Use state-changed messages to
24716 start/stop of scale update timer. Indeed the scale slider was
24717 jumping here and there because the update timer was activated
24718 before seek completed. This fixes instant applying of rate
24720 by pressing the spinbutton like a crazy man !
24722 2006-10-12 19:09:06 +0000 Sebastien Cote <sebas642@yahoo.ca>
24724 gst-libs/gst/rtp/gstbasertppayload.c: Fix two small memory leaks (#361456).
24725 Original commit message from CVS:
24726 Patch by: Sebastien Cote <sebas642 at yahoo.ca>
24727 * gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_init),
24728 (gst_basertppayload_finalize):
24729 Fix two small memory leaks (#361456).
24731 2006-10-10 18:56:01 +0000 Julien Moutte <julien@moutte.net>
24733 tests/examples/seek/seek.c: When changing spinbutton we try to change the rate on the fly.
24734 Original commit message from CVS:
24735 2006-10-10 Julien MOUTTE <julien@moutte.net>
24736 * tests/examples/seek/seek.c: (do_seek),
24737 (rate_spinbutton_changed_cb): When changing spinbutton we try
24738 to change the rate on the fly.
24740 2006-10-10 16:50:06 +0000 Wim Taymans <wim.taymans@gmail.com>
24742 gst-libs/gst/riff/: Add WMS caps.
24743 Original commit message from CVS:
24744 * gst-libs/gst/riff/riff-ids.h:
24745 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps),
24746 (gst_riff_create_audio_template_caps):
24749 2006-10-10 12:49:03 +0000 Josep Torre Valles <josep@fluendo.com>
24751 ext/gnomevfs/: Fix URI interface implementation return type.
24752 Original commit message from CVS:
24753 2006-10-10 Zaheer Abbas Merali <zaheerabbas at merali dot org>
24754 Patch by: Josep Torre Valles <josep@fluendo.com>
24755 * ext/gnomevfs/gstgnomevfssink.c:
24756 * ext/gnomevfs/gstgnomevfssrc.c:
24757 Fix URI interface implementation return type.
24758 * ext/pango/gsttextoverlay.c: (gst_text_overlay_set_property):
24759 Fix what looks like a copy/paste issue when assigning values.
24760 * gst-libs/gst/audio/gstaudiofiltertemplate.c:
24761 (gst_audio_filter_template_get_type):
24762 Cast to prevent Forte warnings.
24763 * gst-libs/gst/cdda/gstcddabasesrc.c: (gst_cdda_base_src_create):
24764 Fix URI interface implementation return type.
24765 gst_pad_query_position requires a signed integer pointer as
24766 3rd parameter, GstClockTime is unsigned.
24767 * gst/audioconvert/audioconvert.c:
24768 Fix integer overflow when treated as signed.
24769 * gst/audioresample/resample.c: (resample_add_input_data):
24770 Cast to prevent warnings on Forte.
24771 * gst/ffmpegcolorspace/imgconvert.c: (build_rgb_palette):
24772 Fix integer overflow when treated as signed.
24773 * gst/ffmpegcolorspace/imgconvert_template.h:
24774 Fix integer overflow when treated as signed. RGBA_OUT shifts bits.
24775 * gst/playback/gstdecodebin.c: (queue_filled_cb),
24776 (cleanup_decodebin):
24777 Who initialises a guint to -1!
24778 Cast function pointers to prevent warnings on Forte.
24779 * gst/playback/gstplaybasebin.c: (queue_deadlock_check),
24780 (queue_threshold_reached):
24781 Cast function pointers correctly to prevent warnings on Forte.
24782 * gst/playback/gststreaminfo.c: (gst_stream_info_dispose):
24783 Cast function pointers correctly to prevent warnings on Forte.
24784 * gst/subparse/gstssaparse.c: (gst_ssa_parse_setcaps):
24785 Obvious change to unsigned, 0xEF > max signed char.
24786 * gst/tcp/gstmultifdsink.c: (get_buffers_max), (count_burst_unit):
24787 GstClockTime is unsigned, initialise correctly.
24788 * gst/tcp/gsttcp.c: (gst_tcp_socket_write):
24789 Cast so pointer arithemetic doesn't cause warnings on Forte.
24790 * gst/videorate/gstvideorate.c:
24791 Use correct return value.
24792 * tests/examples/seek/scrubby.c:
24793 GstClockTime is unsigned, initialise correctly.
24795 2006-10-10 11:20:03 +0000 Ferenc Gerlits <fgerlits@gmail.com>
24797 gst/typefind/gsttypefindfunctions.c: Recognise XML files and XML-like files shorter than 256 bytes as well (fixes #35...
24798 Original commit message from CVS:
24799 Patch by: Ferenc Gerlits <fgerlits at gmail com>
24800 * gst/typefind/gsttypefindfunctions.c:
24801 Recognise XML files and XML-like files shorter than 256 bytes as
24802 well (fixes #359237).
24804 2006-10-09 15:01:30 +0000 Edgard Lima <edgard.lima@indt.org.br>
24808 * gst/typefind/gsttypefindfunctions.c:
24809 Added typefind functions to video/x-nuv media.
24810 Original commit message from CVS:
24811 Added typefind functions to video/x-nuv media.
24813 2006-10-08 16:59:31 +0000 Tim-Philipp Müller <tim@centricular.net>
24815 gst-libs/gst/interfaces/xoverlay.c: Some more guards against invalid input.
24816 Original commit message from CVS:
24817 * gst-libs/gst/interfaces/xoverlay.c:
24818 (gst_x_overlay_set_xwindow_id), (gst_x_overlay_expose):
24819 Some more guards against invalid input.
24821 2006-10-07 18:35:39 +0000 Julien Moutte <julien@moutte.net>
24823 ext/pango/gsttextoverlay.c: Useless goto.
24824 Original commit message from CVS:
24825 2006-10-07 Julien MOUTTE <julien@moutte.net>
24826 * ext/pango/gsttextoverlay.c: (gst_text_overlay_text_event):
24828 * tests/examples/seek/seek.c: (do_seek),
24829 (rate_spinbutton_changed_cb), (main): Add a rate spinbutton in
24830 seek example to experiment with rates != 1.0 (reverse playback
24833 2006-10-06 19:20:53 +0000 Stefan Kost <ensonic@users.sourceforge.net>
24835 gst-libs/gst/interfaces/xoverlay.c: Unref message in doc-example (spotted by Robert McQueen)
24836 Original commit message from CVS:
24837 * gst-libs/gst/interfaces/xoverlay.c:
24838 Unref message in doc-example (spotted by Robert McQueen)
24840 2006-10-06 17:04:05 +0000 Wim Taymans <wim.taymans@gmail.com>
24842 gst/typefind/gsttypefindfunctions.c: printf fix.
24843 Original commit message from CVS:
24844 * gst/typefind/gsttypefindfunctions.c: (mp3_type_find_at_offset),
24845 (mpeg1_parse_header), (mpeg1_sys_type_find):
24848 2006-10-06 14:37:43 +0000 Wim Taymans <wim.taymans@gmail.com>
24850 gst/playback/: Activate dynamic pads before adding them to the element.
24851 Original commit message from CVS:
24852 * gst/playback/gstdecodebin.c: (gst_decode_bin_init),
24854 * gst/playback/gstplaybasebin.c: (new_decoded_pad_full):
24855 Activate dynamic pads before adding them to the element.
24857 2006-10-06 14:04:53 +0000 Michael Smith <msmith@xiph.org>
24859 gst-libs/gst/floatcast/floatcast.h: Fix obviously-bogus macros; use the correct types.
24860 Original commit message from CVS:
24861 * gst-libs/gst/floatcast/floatcast.h:
24862 Fix obviously-bogus macros; use the correct types.
24864 2006-10-06 13:34:46 +0000 Wim Taymans <wim.taymans@gmail.com>
24866 gst-libs/gst/rtp/gstbasertpdepayload.c: Also call parent state change function to activate pads.
24867 Original commit message from CVS:
24868 * gst-libs/gst/rtp/gstbasertpdepayload.c:
24869 (gst_base_rtp_depayload_change_state):
24870 Also call parent state change function to activate pads.
24871 * gst/typefind/gsttypefindfunctions.c: (mp3_type_find_at_offset),
24872 (mpeg1_parse_header), (mpeg1_sys_type_find):
24873 Add some more debug info in mpeg typefinding.
24875 2006-10-06 12:57:10 +0000 Michael Smith <msmith@xiph.org>
24877 ext/theora/theoradec.c: Zero byte theora packets are valid and well-defined; don't warn on them.
24878 Original commit message from CVS:
24879 * ext/theora/theoradec.c: (theora_dec_chain):
24880 Zero byte theora packets are valid and well-defined; don't warn on
24883 2006-10-06 10:04:14 +0000 Thomas Vander Stichele <thomas@apestaart.org>
24885 gst/tcp/gstmultifdsink.c: API: add dropped_buffers to the get-stats GValueArray
24886 Original commit message from CVS:
24887 * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init),
24888 (gst_multi_fd_sink_get_stats), (find_limits),
24889 (gst_multi_fd_sink_queue_buffer):
24890 API: add dropped_buffers to the get-stats GValueArray
24892 2006-10-05 15:55:21 +0000 Tim-Philipp Müller <tim@centricular.net>
24894 Printf format fixes.
24895 Original commit message from CVS:
24896 * ext/alsa/gstalsadeviceprobe.c:
24897 (gst_alsa_device_property_probe_get_values):
24898 * ext/alsa/gstalsasink.c: (set_hwparams):
24899 * ext/ogg/gstoggdemux.c: (gst_ogg_demux_chain_elem_pad),
24900 (gst_ogg_chain_new_stream), (gst_ogg_demux_read_chain):
24901 * ext/ogg/gstoggmux.c: (gst_ogg_mux_send_headers),
24902 (gst_ogg_mux_process_best_pad):
24903 * ext/ogg/gstoggparse.c: (gst_ogg_parse_new_stream),
24904 (gst_ogg_parse_chain):
24905 * ext/ogg/gstogmparse.c: (gst_ogm_parse_stream_header):
24906 * ext/vorbis/vorbisdec.c: (vorbis_handle_data_packet):
24907 * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_setup),
24908 (gst_vorbis_enc_buffer_check_discontinuous):
24909 * ext/vorbis/vorbisparse.c: (vorbis_parse_src_query):
24910 * gst-libs/gst/audio/gstbaseaudiosink.c:
24911 (gst_base_audio_sink_render):
24912 * gst-libs/gst/cdda/gstcddabasesrc.c:
24913 (gst_cdda_base_src_handle_track_seek):
24914 * gst-libs/gst/rtp/gstbasertpdepayload.c:
24915 (gst_base_rtp_depayload_push_full):
24916 * gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_push):
24917 * gst/audioresample/resample.c: (resample_input_pushthrough):
24918 * gst/playback/gstplaybasebin.c: (queue_out_of_data):
24919 * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_handle_clients):
24920 * gst/typefind/gsttypefindfunctions.c: (mp3_type_find_at_offset),
24921 (wavpack_type_find):
24922 * gst/videotestsrc/gstvideotestsrc.c: (gst_video_test_src_create):
24923 * sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_destroy),
24924 (gst_xvimagesink_check_xshm_calls), (gst_xvimagesink_xvimage_new):
24925 * tests/check/elements/volume.c: (GST_START_TEST):
24926 Printf format fixes.
24928 2006-10-04 13:18:57 +0000 Thomas Vander Stichele <thomas@apestaart.org>
24930 gst/tcp/gsttcp.c: Fix a simple mistake (see the docs)
24931 Original commit message from CVS:
24932 * gst/tcp/gsttcp.c: (gst_tcp_gdp_read_caps):
24933 Fix a simple mistake (see the docs)
24936 2006-10-04 13:15:36 +0000 Thomas Vander Stichele <thomas@apestaart.org>
24938 * win32/common/config.h:
24940 Original commit message from CVS:
24943 2006-10-03 12:11:45 +0000 Tim-Philipp Müller <tim@centricular.net>
24945 docs/plugins/: Add vorbistag element to docs; update version numbers to 0.10.10.1.
24946 Original commit message from CVS:
24947 * docs/plugins/Makefile.am:
24948 * docs/plugins/gst-plugins-base-plugins-docs.sgml:
24949 * docs/plugins/gst-plugins-base-plugins-sections.txt:
24950 * docs/plugins/gst-plugins-base-plugins.args:
24951 * docs/plugins/gst-plugins-base-plugins.hierarchy:
24952 * docs/plugins/inspect/plugin-adder.xml:
24953 * docs/plugins/inspect/plugin-alsa.xml:
24954 * docs/plugins/inspect/plugin-audioconvert.xml:
24955 * docs/plugins/inspect/plugin-audiorate.xml:
24956 * docs/plugins/inspect/plugin-audioresample.xml:
24957 * docs/plugins/inspect/plugin-audiotestsrc.xml:
24958 * docs/plugins/inspect/plugin-cdparanoia.xml:
24959 * docs/plugins/inspect/plugin-decodebin.xml:
24960 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
24961 * docs/plugins/inspect/plugin-gdp.xml:
24962 * docs/plugins/inspect/plugin-gnomevfs.xml:
24963 * docs/plugins/inspect/plugin-libvisual.xml:
24964 * docs/plugins/inspect/plugin-ogg.xml:
24965 * docs/plugins/inspect/plugin-pango.xml:
24966 * docs/plugins/inspect/plugin-playbin.xml:
24967 * docs/plugins/inspect/plugin-subparse.xml:
24968 * docs/plugins/inspect/plugin-tcp.xml:
24969 * docs/plugins/inspect/plugin-theora.xml:
24970 * docs/plugins/inspect/plugin-typefindfunctions.xml:
24971 * docs/plugins/inspect/plugin-video4linux.xml:
24972 * docs/plugins/inspect/plugin-videorate.xml:
24973 * docs/plugins/inspect/plugin-videoscale.xml:
24974 * docs/plugins/inspect/plugin-videotestsrc.xml:
24975 * docs/plugins/inspect/plugin-volume.xml:
24976 * docs/plugins/inspect/plugin-vorbis.xml:
24977 * docs/plugins/inspect/plugin-ximagesink.xml:
24978 * docs/plugins/inspect/plugin-xvimagesink.xml:
24979 Add vorbistag element to docs; update version numbers to 0.10.10.1.
24981 2006-10-03 11:51:48 +0000 James Doc Livingston <doclivingston@gmail.com>
24983 ext/vorbis/: Add new vorbistag element which derives from vorbisparse and is essentially the same as well, only that ...
24984 Original commit message from CVS:
24985 Patch by: James "Doc" Livingston <doclivingston at gmail com>
24986 * ext/vorbis/Makefile.am:
24987 * ext/vorbis/vorbis.c: (plugin_init):
24988 * ext/vorbis/vorbisparse.c: (gst_vorbis_parse_class_init),
24989 (vorbis_parse_parse_packet), (vorbis_parse_chain):
24990 * ext/vorbis/vorbisparse.h:
24991 * ext/vorbis/vorbistag.c: (gst_vorbis_tag_base_init),
24992 (gst_vorbis_tag_class_init), (gst_vorbis_tag_init),
24993 (gst_vorbis_tag_parse_packet):
24994 * ext/vorbis/vorbistag.h:
24995 Add new vorbistag element which derives from vorbisparse
24996 and is essentially the same as well, only that it implements
24997 the GstTagSetter interface and can modify the stream's
24998 vorbiscomment on the fly (#335635).
24999 * tests/check/Makefile.am:
25000 * tests/check/elements/.cvsignore:
25001 * tests/check/elements/vorbistag.c: (setup_vorbistag),
25002 (cleanup_vorbistag), (buffer_probe), (start_pipeline),
25003 (get_buffer), (stop_pipeline), (_create_codebook_header_buffer),
25004 (_create_audio_buffer), (GST_START_TEST), (vorbistag_suite):
25005 Add unit test for new vorbistag element.
25007 2006-10-03 10:36:38 +0000 Tim-Philipp Müller <tim@centricular.net>
25009 ext/vorbis/vorbisparse.c: Set BOS flag in packet structure to fix 'jump depends on unitialized value' errors in valgr...
25010 Original commit message from CVS:
25011 * ext/vorbis/vorbisparse.c: (gst_vorbis_parse_init),
25012 (vorbis_parse_push_headers), (vorbis_parse_chain):
25013 Set BOS flag in packet structure to fix 'jump depends
25014 on unitialized value' errors in valgrind; various minor
25017 2006-09-30 15:30:07 +0000 Jan Schmidt <thaytan@mad.scientist.com>
25019 gst/playback/gstdecodebin.c: Fix typo in a debug statement.
25020 Original commit message from CVS:
25021 * gst/playback/gstdecodebin.c: (close_pad_link):
25022 Fix typo in a debug statement.
25023 * gst/playback/gstplaybasebin.c: (probe_triggered),
25024 (new_decoded_pad_full), (new_decoded_pad), (subs_new_decoded_pad),
25025 (gen_source_element), (source_new_pad), (analyse_source),
25027 When handling no_more_pads in new_decoded_pad, make sure to treat
25028 subtitle pads correctly. Fixes playback with subtitle files.
25029 Move a recurring message to LOG level.
25030 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support):
25031 The maximum value for the Xv colorkey on this Radeon is 0xFFFFFFFF,
25032 which ends up as -1 when cast to an int. Make the logic handle the
25033 max value as an unsigned mask and only change the colorkey when it's
25034 a value we recognise.
25036 2006-09-30 00:14:20 +0000 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
25038 gst-libs/gst/rtp/gstbasertpaudiopayload.c: Removed empty * between paragraphs
25039 Original commit message from CVS:
25040 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
25041 Removed empty * between paragraphs
25043 2006-09-29 23:50:53 +0000 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
25045 gst-libs/gst/rtp/: Moved some documentation into .c file
25046 Original commit message from CVS:
25047 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
25048 * gst-libs/gst/rtp/README:
25049 Moved some documentation into .c file
25051 2006-09-29 17:35:01 +0000 Wim Taymans <wim.taymans@gmail.com>
25053 gst/playback/gstdecodebin.c: Fix compilation.
25054 Original commit message from CVS:
25055 * gst/playback/gstdecodebin.c: (no_more_pads):
25058 2006-09-29 16:04:05 +0000 Wim Taymans <wim.taymans@gmail.com>
25060 gst/playback/gstdecodebin.c: Remove g_print
25061 Original commit message from CVS:
25062 * gst/playback/gstdecodebin.c: (new_caps):
25064 * gst/playback/gstplaybin.c:
25067 2006-09-29 15:16:32 +0000 Tim-Philipp Müller <tim@centricular.net>
25069 tests/check/Makefile.am: Re-enable cddabasesrc test to see if it works again now.
25070 Original commit message from CVS:
25071 * tests/check/Makefile.am:
25072 Re-enable cddabasesrc test to see if it works again
25075 2006-09-29 13:46:45 +0000 Wim Taymans <wim.taymans@gmail.com>
25077 gst/playback/gstplaybasebin.c: Handle invalid URIs a bit more gracefully.
25078 Original commit message from CVS:
25079 * gst/playback/gstplaybasebin.c: (setup_subtitle),
25080 (gen_source_element):
25081 Handle invalid URIs a bit more gracefully.
25083 2006-09-29 12:54:28 +0000 Tim-Philipp Müller <tim@centricular.net>
25085 tests/check/pipelines/oggmux.c: Remove obsolete comment.
25086 Original commit message from CVS:
25087 * tests/check/pipelines/oggmux.c:
25088 Remove obsolete comment.
25090 2006-09-29 10:43:05 +0000 James Doc Livingston <doclivingston@gmail.com>
25092 ext/ogg/gstoggmux.c: Commit patch from James "Doc" Livingston, adds proper EOS handling in oggmux. GStreamer can, for...
25093 Original commit message from CVS:
25094 * ext/ogg/gstoggmux.c: (gst_ogg_mux_request_new_pad),
25095 (gst_ogg_mux_release_pad), (gst_ogg_mux_push_buffer),
25096 (gst_ogg_mux_compare_pads), (gst_ogg_mux_queue_pads),
25097 (gst_ogg_mux_send_headers), (gst_ogg_mux_process_best_pad),
25098 (gst_ogg_mux_collected):
25099 Commit patch from James "Doc" Livingston, adds proper EOS handling
25100 in oggmux. GStreamer can, for the first time ever, create a valid
25102 * tests/check/pipelines/oggmux.c: (check_chain_final_state),
25104 Reenable tests now that they pass.
25106 2006-09-29 08:20:03 +0000 Wim Taymans <wim.taymans@gmail.com>
25108 gst/tcp/gstmultifdsink.c: Stop reading commands when EOF (we read 0) as well.
25109 Original commit message from CVS:
25110 * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_handle_clients):
25111 Stop reading commands when EOF (we read 0) as well.
25113 2006-09-28 15:29:17 +0000 Wim Taymans <wim.taymans@gmail.com>
25115 gst/playback/gstdecodebin.c: Implement delayed caps linking needed for element with a lot of different caps on the sr...
25116 Original commit message from CVS:
25117 * gst/playback/gstdecodebin.c: (dynamic_create), (dynamic_free),
25118 (close_pad_link), (dynamic_remove), (no_more_pads), (new_caps),
25119 (find_dynamic), (unlinked), (close_link):
25120 Implement delayed caps linking needed for element with a lot of
25121 different caps on the src pads that get fixed at runtime.
25122 Improve management of dynamic elements.
25123 * gst/playback/gstplaybasebin.c: (gst_play_base_bin_init),
25124 (group_destroy), (group_commit), (check_queue), (queue_overrun),
25125 (gen_preroll_element), (remove_groups), (unknown_type),
25126 (add_element_stream), (no_more_pads_full), (no_more_pads),
25127 (sub_no_more_pads), (source_no_more_pads), (preroll_unlinked),
25128 (new_decoded_pad), (setup_subtitle), (array_has_value),
25129 (gen_source_element), (source_new_pad), (has_all_raw_caps),
25130 (analyse_source), (remove_decoders), (make_decoder),
25131 (remove_source), (setup_source), (finish_source), (prepare_output),
25132 (gst_play_base_bin_change_state):
25133 * gst/playback/gstplaybasebin.h:
25134 Use more _CAST instead of full type checking casts.
25135 Small cleanups, plug some leaks.
25136 Handle dynamic sources.
25137 Add some helper functions to create lists of strings used for
25138 blacklisting and other stuff.
25139 Refactor some code dealing with analysing the source.
25140 Re-enable sources without pads (like cd:// or other selfcontained
25143 2006-09-28 15:08:15 +0000 Wim Taymans <wim.taymans@gmail.com>
25145 gst-libs/gst/audio/gstbaseaudiosink.c: When we have a timestamp, we can still perform clipping.
25146 Original commit message from CVS:
25147 * gst-libs/gst/audio/gstbaseaudiosink.c:
25148 (gst_base_audio_sink_render):
25149 When we have a timestamp, we can still perform clipping.
25150 When we have no clock, we must play the sample ASAP.
25152 2006-09-28 11:46:26 +0000 Wim Taymans <wim.taymans@gmail.com>
25154 gst/audiorate/gstaudiorate.c: Set caps on outgoing buffers.
25155 Original commit message from CVS:
25156 * gst/audiorate/gstaudiorate.c: (gst_audio_rate_chain):
25157 Set caps on outgoing buffers.
25158 * gst/videorate/gstvideorate.c: (gst_video_rate_flush_prev),
25159 (gst_video_rate_event), (gst_video_rate_chain):
25160 * gst/videorate/gstvideorate.h:
25161 Fix videorate some more. Fixes #357977
25163 2006-09-28 11:34:05 +0000 Tim-Philipp Müller <tim@centricular.net>
25165 tests/check/elements/adder.c: Don't set timeout to 6 seconds when we're running in valgrind ... (and how is 6 seconds...
25166 Original commit message from CVS:
25167 * tests/check/elements/adder.c: (adder_suite):
25168 Don't set timeout to 6 seconds when we're running
25169 in valgrind ... (and how is 6 seconds longer than
25170 the default anyway?)
25172 2006-09-28 10:49:56 +0000 Wim Taymans <wim.taymans@gmail.com>
25174 gst/audiorate/gstaudiorate.c: Keep sink and src segment to keep track of time and support more input formats.
25175 Original commit message from CVS:
25176 * gst/audiorate/gstaudiorate.c: (gst_audio_rate_reset),
25177 (gst_audio_rate_sink_event), (gst_audio_rate_convert),
25178 (gst_audio_rate_convert_segments), (gst_audio_rate_chain):
25179 Keep sink and src segment to keep track of time and support more
25181 Fix bogus next_offset and run_time calculation, don't understand how
25182 this could have worked before. Fixes #357976.
25183 Remove some unneeded vars.
25185 2006-09-28 09:41:20 +0000 Tim-Philipp Müller <tim@centricular.net>
25187 gst/playback/gstplaybin.c: Only remove visualisation from visbin if there is a visbin (or: don't throw warnings when ...
25188 Original commit message from CVS:
25189 * gst/playback/gstplaybin.c: (remove_sinks):
25190 Only remove visualisation from visbin if there is a visbin (or:
25191 don't throw warnings when closing totem without playing a file).
25193 2006-09-27 13:52:14 +0000 Wim Taymans <wim.taymans@gmail.com>
25195 gst-libs/gst/audio/gstbaseaudiosink.c: Add some more info in a WARNING.
25196 Original commit message from CVS:
25197 * gst-libs/gst/audio/gstbaseaudiosink.c:
25198 (gst_base_audio_sink_render):
25199 Add some more info in a WARNING.
25200 * gst-libs/gst/audio/gstbaseaudiosrc.c:
25201 (gst_base_audio_src_create):
25202 Handle PAUSE in create function, use new -core addition to
25203 wait for playing. Fixes pausing and resuming capture from an
25205 * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_commit),
25206 (gst_ring_buffer_read):
25207 Constify some more.
25208 Caller supports interrupted reads now.
25210 2006-09-27 13:29:49 +0000 Christian Schaller <uraeus@gnome.org>
25212 * gst-plugins-base.spec.in:
25213 add new header file to spec
25214 Original commit message from CVS:
25215 add new header file to spec
25217 2006-09-27 12:55:45 +0000 Tim-Philipp Müller <tim@centricular.net>
25219 tests/check/Makefile.am: Another attempt to make the gen64 buildbot happy.
25220 Original commit message from CVS:
25221 * tests/check/Makefile.am:
25222 Another attempt to make the gen64 buildbot happy.
25224 2006-09-27 11:58:17 +0000 Jonathan Matthew <jonathan@kaolin.wh9.net>
25226 ext/libvisual/visual.c: Libvisual plugin was not passing audio data to libvisual 0.4.0 correctly. Fixes #357800
25227 Original commit message from CVS:
25228 Patch by: Jonathan Matthew <jonathan@kaolin.wh9.net>
25229 * ext/libvisual/visual.c: (gst_visual_clear_actors),
25230 (gst_visual_chain), (gst_visual_change_state):
25231 Libvisual plugin was not passing audio data to libvisual 0.4.0
25232 correctly. Fixes #357800
25234 2006-09-27 11:31:43 +0000 Tim-Philipp Müller <tim@centricular.net>
25236 tests/check/pipelines/simple-launch-lines.c: Add timeout to _get_state() so we see which pipeline it is that causes t...
25237 Original commit message from CVS:
25238 * tests/check/pipelines/simple-launch-lines.c: (run_pipeline):
25239 Add timeout to _get_state() so we see which pipeline it is
25240 that causes trouble on the gen64 build bot.
25242 2006-09-27 11:06:54 +0000 Wim Taymans <wim.taymans@gmail.com>
25244 gst-libs/gst/rtp/gstbasertpdepayload.c: the source pad always uses fixed caps.
25245 Original commit message from CVS:
25246 * gst-libs/gst/rtp/gstbasertpdepayload.c:
25247 (gst_base_rtp_depayload_init), (gst_base_rtp_depayload_push_full),
25248 (gst_base_rtp_depayload_push_ts), (gst_base_rtp_depayload_process),
25249 (gst_base_rtp_depayload_set_gst_timestamp):
25250 the source pad always uses fixed caps.
25252 2006-09-27 11:05:08 +0000 Wim Taymans <wim.taymans@gmail.com>
25254 Added docs for the audio libs.
25255 Original commit message from CVS:
25256 * docs/libs/gst-plugins-base-libs-docs.sgml:
25257 * docs/libs/gst-plugins-base-libs-sections.txt:
25258 * gst-libs/gst/audio/gstaudioclock.c:
25259 * gst-libs/gst/audio/gstaudioclock.h:
25260 * gst-libs/gst/audio/gstaudiosink.c:
25261 * gst-libs/gst/audio/gstaudiosink.h:
25262 * gst-libs/gst/audio/gstaudiosrc.c:
25263 * gst-libs/gst/audio/gstbaseaudiosink.c:
25264 (gst_base_audio_sink_render):
25265 * gst-libs/gst/audio/gstbaseaudiosink.h:
25266 * gst-libs/gst/audio/gstbaseaudiosrc.c: (gst_base_audio_src_init):
25267 * gst-libs/gst/audio/gstbaseaudiosrc.h:
25268 * gst-libs/gst/audio/gstringbuffer.h:
25269 Added docs for the audio libs.
25271 2006-09-27 10:59:24 +0000 Tim-Philipp Müller <tim@centricular.net>
25273 tests/check/Makefile.am: Temporarily disable test that fails on the bots for unknown reasons.
25274 Original commit message from CVS:
25275 * tests/check/Makefile.am:
25276 Temporarily disable test that fails on the bots for unknown reasons.
25278 2006-09-27 00:13:29 +0000 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
25280 gst-libs/gst/rtp/gstbasertpaudiopayload.*: Moved AudioCodecType into priv
25281 Original commit message from CVS:
25282 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
25283 * gst-libs/gst/rtp/gstbasertpaudiopayload.h:
25284 Moved AudioCodecType into priv
25285 Renamed all gst_basertpaudiopayload to gst_base_rtp_audio_payload prefixes
25287 2006-09-25 15:47:25 +0000 Wim Taymans <wim.taymans@gmail.com>
25289 gst/playback/gstdecodebin.c: Cleanups and small leak fixes.
25290 Original commit message from CVS:
25291 * gst/playback/gstdecodebin.c: (gst_decode_bin_factory_filter),
25292 (add_fakesink), (remove_fakesink), (pad_probe), (close_pad_link),
25293 (is_demuxer_element), (try_to_link_1), (get_our_ghost_pad),
25295 Cleanups and small leak fixes.
25296 Added Depayloaders to valid list of autopluggable elements.
25298 2006-09-25 13:24:59 +0000 Wim Taymans <wim.taymans@gmail.com>
25300 gst/playback/gstplaybin.c: Detect NO_PREROLL state change returns and disable clock distribution to the sinks so that...
25301 Original commit message from CVS:
25302 * gst/playback/gstplaybin.c: (gst_play_bin_class_init),
25303 (gst_play_bin_vis_blocked), (gst_play_bin_set_property),
25304 (gen_video_element), (gen_text_element), (gen_audio_element),
25305 (gen_vis_element), (remove_sinks), (add_sink), (setup_sinks),
25306 (gst_play_bin_set_clock_func), (gst_play_bin_change_state):
25307 Detect NO_PREROLL state change returns and disable clock distribution to
25308 the sinks so that sync is disabled.
25309 Avoid some type checking and do simple casts instead.
25310 Small cleanups, fix some FIXMEs.
25311 Be more robust when linking user specified elements, catch an report
25312 errors. Fixes #357404.
25313 Fix some leaks in the error paths.
25315 2006-09-25 12:55:04 +0000 Stefan Kost <ensonic@users.sourceforge.net>
25318 ChangeLog surgery for missing bug-number
25319 Original commit message from CVS:
25320 ChangeLog surgery for missing bug-number
25322 2006-09-25 11:28:15 +0000 Peter Kjellerstedt <pkj@axis.com>
25324 gst/playback/test.c: Fix compilation with uClibc and -Werror (#357591).
25325 Original commit message from CVS:
25326 Patch by: Peter Kjellerstedt <pkj at axis com>
25327 * gst/playback/test.c:
25328 Fix compilation with uClibc and -Werror (#357591).
25330 2006-09-25 10:21:31 +0000 Tim-Philipp Müller <tim@centricular.net>
25332 gst-libs/gst/tag/gstvorbistag.c: Parse dates that are followed by a time as well (#357532).
25333 Original commit message from CVS:
25334 * gst-libs/gst/tag/gstvorbistag.c: (gst_vorbis_tag_add):
25335 Parse dates that are followed by a time as well (#357532).
25336 * tests/check/libs/tag.c: (test_vorbis_tags):
25337 Add unit test for this.
25339 2006-09-23 15:24:55 +0000 Tim-Philipp Müller <tim@centricular.net>
25341 gst/: A few array const-ifications.
25342 Original commit message from CVS:
25343 * gst/audioconvert/gstaudioconvert.c: (make_lossless_changes),
25344 (gst_audio_convert_transform_caps):
25345 * gst/videotestsrc/videotestsrc.c: (gst_video_test_src_unicolor):
25346 * gst/videotestsrc/videotestsrc.h:
25347 A few array const-ifications.
25349 2006-09-23 15:02:51 +0000 Tim-Philipp Müller <tim@centricular.net>
25351 tests/check/Makefile.am: See if this makes the build bots happy.
25352 Original commit message from CVS:
25353 * tests/check/Makefile.am:
25354 See if this makes the build bots happy.
25355 * tests/check/libs/cddabasesrc.c:
25358 2006-09-23 14:30:53 +0000 Young-Ho Cha <ganadist@chollian.net>
25360 gst/subparse/samiparse.c: More case-insensitivity for certain tags; recognise entities with decimal codes as special ...
25361 Original commit message from CVS:
25362 Patch by: Young-Ho Cha <ganadist at chollian dot net>
25363 * gst/subparse/samiparse.c: (handle_start_font),
25364 (fix_invalid_entities):
25365 More case-insensitivity for certain tags; recognise entities with
25366 decimal codes as special entities as well (#357330).
25368 2006-09-23 13:32:07 +0000 Tim-Philipp Müller <tim@centricular.net>
25370 gst-libs/gst/Makefile.am: Need to build tag directory before cdda.
25371 Original commit message from CVS:
25372 * gst-libs/gst/Makefile.am:
25373 Need to build tag directory before cdda.
25375 2006-09-23 13:21:07 +0000 Tim-Philipp Müller <tim@centricular.net>
25377 Move GST_TAG_CDDA_* tags into libgsttag and make libgstcddabasesrc depend on libgsttag. This is required so we can ex...
25378 Original commit message from CVS:
25379 * docs/libs/gst-plugins-base-libs-sections.txt:
25380 * gst-libs/gst/cdda/Makefile.am:
25381 * gst-libs/gst/cdda/gstcddabasesrc.c:
25382 (gst_cdda_base_src_base_init):
25383 * gst-libs/gst/cdda/gstcddabasesrc.h:
25384 * gst-libs/gst/tag/tag.h:
25385 * gst-libs/gst/tag/tags.c: (gst_tag_register_tags_internal),
25386 (gst_tag_register_musicbrainz_tags):
25387 Move GST_TAG_CDDA_* tags into libgsttag and make libgstcddabasesrc
25388 depend on libgsttag. This is required so we can extract/read tags like
25389 DISCID without depending on libgstcddabasesrc (which used to register
25391 * gst-libs/gst/tag/gstvorbistag.c:
25392 Add vorbiscomment mapping for CDDB_DISCID and MUSICBRAINZ_DISCID
25393 tags (also see #347848).
25394 * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_metadata_set1):
25395 Log vorbis comments we are actually writing. Const-ify array.
25397 2006-09-23 08:53:30 +0000 Wim Taymans <wim.taymans@gmail.com>
25399 gst/playback/gstplaybasebin.c: Improve buffering a bit by avoiding a deadlock because we cannot assume the underrun i...
25400 Original commit message from CVS:
25401 * gst/playback/gstplaybasebin.c: (gen_preroll_element):
25402 Improve buffering a bit by avoiding a deadlock because we cannot assume
25403 the underrun is always called.
25405 2006-09-23 08:51:14 +0000 Young-Ho Cha <ganadist@chollian.net>
25407 gst-libs/gst/riff/: Added MPEG-4 AAC and id and caps. Fixes #357289
25408 Original commit message from CVS:
25409 Patch by: Young-Ho Cha <ganadist at chollian dot net>
25410 * gst-libs/gst/riff/riff-ids.h:
25411 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps),
25412 (gst_riff_create_audio_template_caps):
25413 Added MPEG-4 AAC and id and caps. Fixes #357289
25414 Added WMA9 Lossless id.
25416 2006-09-22 14:50:01 +0000 Tim-Philipp Müller <tim@centricular.net>
25418 ext/gnomevfs/gstgnomevfssrc.c: Fix misleading docs addition.
25419 Original commit message from CVS:
25420 * ext/gnomevfs/gstgnomevfssrc.c:
25421 Fix misleading docs addition.
25422 * tests/check/elements/videotestsrc.c: (check_rgb_buf):
25423 Get rid of compiler warning the right way.
25425 2006-09-22 14:13:34 +0000 Wim Taymans <wim.taymans@gmail.com>
25427 gst-libs/gst/rtp/gstbasertpdepayload.*: Small cleanups.
25428 Original commit message from CVS:
25429 * gst-libs/gst/rtp/gstbasertpdepayload.c:
25430 (gst_base_rtp_depayload_finalize),
25431 (gst_base_rtp_depayload_setcaps), (gst_base_rtp_depayload_chain),
25432 (gst_base_rtp_depayload_push_full),
25433 (gst_base_rtp_depayload_push_ts), (gst_base_rtp_depayload_push),
25434 (gst_base_rtp_depayload_process),
25435 (gst_base_rtp_depayload_set_gst_timestamp),
25436 (gst_base_rtp_depayload_queue_release):
25437 * gst-libs/gst/rtp/gstbasertpdepayload.h:
25440 Refactored the process method and added methods to push from the process
25442 Use _scale functions.
25443 API: gst_base_rtp_depayload_push_ts
25444 API: gst_base_rtp_depayload_push
25445 * gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_push):
25446 timestamps are uint.
25448 2006-09-22 11:59:00 +0000 Stefan Kost <ensonic@users.sourceforge.net>
25450 gst-libs/gst/interfaces/xoverlay.c: Remove unused statement from doc example.
25451 Original commit message from CVS:
25452 * gst-libs/gst/interfaces/xoverlay.c:
25453 Remove unused statement from doc example.
25455 2006-09-22 09:52:21 +0000 Thomas Vander Stichele <thomas@apestaart.org>
25457 * gst/videorate/gstvideorate.c:
25459 Original commit message from CVS:
25462 2006-09-21 13:49:47 +0000 Stefan Kost <ensonic@users.sourceforge.net>
25464 gst-libs/gst/interfaces/videoorientation.c: Add since tags to new API docs, ChangeLog surgery (forgot API keyword in ...
25465 Original commit message from CVS:
25466 * gst-libs/gst/interfaces/videoorientation.c:
25467 (gst_video_orientation_iface_init),
25468 (gst_video_orientation_get_hflip),
25469 (gst_video_orientation_get_vflip),
25470 (gst_video_orientation_get_hcenter),
25471 (gst_video_orientation_get_vcenter),
25472 (gst_video_orientation_set_hflip),
25473 (gst_video_orientation_set_vflip),
25474 (gst_video_orientation_set_hcenter),
25475 (gst_video_orientation_set_vcenter):
25476 Add since tags to new API docs, ChangeLog surgery (forgot API keyword
25479 2006-09-21 09:27:47 +0000 Tim-Philipp Müller <tim@centricular.net>
25481 tests/check/: but disable for now since it doesn't pass (something wrong with
25482 Original commit message from CVS:
25483 * tests/check/Makefile.am:
25484 * tests/check/elements/.cvsignore:
25485 * tests/check/elements/ffmpegcolorspace.c: (rgb_format_to_caps),
25486 (create_rgb_conversions), (rgb_conversion_free),
25487 (right_shift_colour), (fix_expected_colour), (check_rgb_buf),
25488 (got_buf_cb), (GST_START_TEST), (ffmpegcolorspace_suite):
25489 Add unit test for ffmpegcolorspace (RGB <=> RGB only so far),
25490 but disable for now since it doesn't pass (something wrong with
25493 2006-09-21 07:01:48 +0000 Wim Taymans <wim.taymans@gmail.com>
25495 gst/playback/gstplaybasebin.c: Refactor handling of overrun detection.
25496 Original commit message from CVS:
25497 * gst/playback/gstplaybasebin.c: (group_commit),
25498 (queue_deadlock_check), (queue_overrun), (queue_threshold_reached),
25499 (queue_out_of_data), (gen_preroll_element),
25500 (preroll_remove_overrun), (probe_triggered):
25501 Refactor handling of overrun detection.
25502 Separate handling of group completion and deadlock detection when doing
25503 network buffering. This should fix some deadlocks that were not detected
25504 because the group was completed.
25505 Add more comments, improve debugging.
25507 2006-09-21 05:31:00 +0000 Wim Taymans <wim.taymans@gmail.com>
25509 tests/check/: Some more compilation fixes.
25510 Original commit message from CVS:
25511 * tests/check/elements/gdpdepay.c: (GST_START_TEST):
25512 * tests/check/libs/audio.c:
25513 Some more compilation fixes.
25515 2006-09-21 05:12:18 +0000 Wim Taymans <wim.taymans@gmail.com>
25517 gst-libs/gst/audio/gstringbuffer.c: Early morning compilation fix.
25518 Original commit message from CVS:
25519 * gst-libs/gst/audio/gstringbuffer.c:
25520 (gst_ring_buffer_samples_done), (gst_ring_buffer_commit),
25521 (gst_ring_buffer_read):
25522 Early morning compilation fix.
25524 2006-09-20 18:09:43 +0000 Thomas Vander Stichele <thomas@apestaart.org>
25528 Original commit message from CVS:
25531 2006-09-20 17:04:57 +0000 Wim Taymans <wim.taymans@gmail.com>
25533 tests/check/: Fix some warnings.
25534 Original commit message from CVS:
25535 * tests/check/elements/gdpdepay.c: (GST_START_TEST):
25536 * tests/check/elements/multifdsink.c: (GST_START_TEST):
25537 * tests/check/elements/videorate.c: (GST_START_TEST):
25538 * tests/check/libs/cddabasesrc.c: (GST_START_TEST):
25539 * tests/check/pipelines/oggmux.c: (eos_buffer_probe):
25542 2006-09-20 10:59:06 +0000 Stefan Kost <ensonic@users.sourceforge.net>
25544 sys/xvimage/xvimagesink.c: change colorkey behaviour back according to #354773 comment 6/7
25545 Original commit message from CVS:
25546 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support),
25547 (gst_xvimagesink_get_times):
25548 change colorkey behaviour back according to #354773 comment 6/7
25550 2006-09-20 10:42:34 +0000 Tim-Philipp Müller <tim@centricular.net>
25553 ChangeLog surgery: remove junk
25554 Original commit message from CVS:
25555 ChangeLog surgery: remove junk
25557 2006-09-19 11:31:06 +0000 Michael Smith <msmith@xiph.org>
25559 gst/tcp/gstmultifdsink.*: Implement stubbed out properties unit-type, units-soft-max, units-max, to allow specifying ...
25560 Original commit message from CVS:
25561 * gst/tcp/gstmultifdsink.c: (gst_sync_method_get_type),
25562 (gst_multi_fd_sink_class_init), (get_buffers_max), (find_limits),
25563 (gst_multi_fd_sink_recover_client),
25564 (gst_multi_fd_sink_queue_buffer), (gst_multi_fd_sink_set_property),
25565 (gst_multi_fd_sink_get_property):
25566 * gst/tcp/gstmultifdsink.h:
25567 Implement stubbed out properties unit-type, units-soft-max,
25568 units-max, to allow specifying maximum sizes in units other than
25572 2006-09-19 10:23:03 +0000 Wim Taymans <wim.taymans@gmail.com>
25574 gst-libs/gst/riff/riff-media.c: Reorder the audio formats a bit for clarity.
25575 Original commit message from CVS:
25576 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps),
25577 (gst_riff_create_audio_template_caps):
25578 Reorder the audio formats a bit for clarity.
25579 Detect and create caps for MSGSM and MSN (WAV49).
25581 * sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_destroy),
25582 (gst_xvimagesink_check_xshm_calls), (gst_xvimagesink_xvimage_new),
25583 (gst_xvimagesink_get_xv_support), (gst_xvimagesink_show_frame):
25584 Small cleanups, move error handling out of normal flow for clarity.
25586 2006-09-18 15:59:39 +0000 Stefan Kost <ensonic@users.sourceforge.net>
25588 Add new interface to control video orientation (fixes #354908)
25589 Original commit message from CVS:
25590 * docs/libs/gst-plugins-base-libs-docs.sgml:
25591 * docs/libs/gst-plugins-base-libs.types:
25592 * gst-libs/gst/interfaces/Makefile.am:
25593 * gst-libs/gst/interfaces/videoorientation.c:
25594 (gst_video_orientation_get_type),
25595 (gst_video_orientation_iface_init),
25596 (gst_video_orientation_get_hflip),
25597 (gst_video_orientation_get_vflip),
25598 (gst_video_orientation_get_hcenter),
25599 (gst_video_orientation_get_vcenter),
25600 (gst_video_orientation_set_hflip),
25601 (gst_video_orientation_set_vflip),
25602 (gst_video_orientation_set_hcenter),
25603 (gst_video_orientation_set_vcenter):
25604 * gst-libs/gst/interfaces/videoorientation.h:
25605 Add new interface to control video orientation (fixes #354908)
25607 2006-09-18 15:48:01 +0000 Stefan Kost <ensonic@users.sourceforge.net>
25609 gst/videotestsrc/gstvideotestsrc.c: Use G_UNLIKELY in _create and log one more detail.
25610 Original commit message from CVS:
25611 * gst/videotestsrc/gstvideotestsrc.c:
25612 Use G_UNLIKELY in _create and log one more detail.
25613 (gst_video_test_src_get_times), (gst_video_test_src_create):
25614 * sys/ximage/ximagesink.c: (gst_ximagesink_get_times):
25615 Use gst_util_uint64_scale_int in _get_times().
25617 2006-09-18 15:00:51 +0000 Stefan Kost <ensonic@users.sourceforge.net>
25619 sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support)
25620 Original commit message from CVS:
25621 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support)
25622 Give better warning message (add object and detail).
25624 2006-09-18 14:42:03 +0000 Stefan Kost <ensonic@users.sourceforge.net>
25626 sys/xvimage/xvimagesink.c: xvimage assumed that XV_COLORKEY can be set in RGB888 format (fixes #354773), use gst_util...
25627 Original commit message from CVS:
25628 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support),
25629 (gst_xvimagesink_get_times):
25630 xvimage assumed that XV_COLORKEY can be set in RGB888 format (fixes
25631 #354773), use gst_util_uint64_scale_int in _get_times()
25633 2006-09-18 14:21:45 +0000 Michael Smith <msmith@xiph.org>
25635 ext/ogg/gstoggmux.c: Timestamps are unsigned; comparision against GST_CLOCK_TIME_NONE was always true, leading to dro...
25636 Original commit message from CVS:
25637 * ext/ogg/gstoggmux.c: (gst_ogg_mux_push_buffer):
25638 Timestamps are unsigned; comparision against GST_CLOCK_TIME_NONE was
25639 always true, leading to dropping all timestamps.
25641 2006-09-18 11:40:14 +0000 Stefan Kost <ensonic@users.sourceforge.net>
25643 ext/libvisual/visual.c: update to work also with libvisual 0.4 API
25644 Original commit message from CVS:
25645 * ext/libvisual/visual.c: (gst_vis_src_negotiate),
25646 (gst_visual_chain), (gst_visual_change_state):
25647 update to work also with libvisual 0.4 API
25648 * tools/gst-launch-ext.1.in:
25649 * tools/gst-visualise.1.in:
25650 remove references to old man-pages
25651 * tests/examples/seek/seek.c: (main):
25652 add real meadi-buttons, add tool-tips for the seek-options, arrange
25653 seek options in a table
25655 2006-09-18 10:57:28 +0000 Michael Smith <msmith@xiph.org>
25657 ext/ogg/gstoggmux.c: Don't generate out-of-order timestamps from oggmux, instead clamp output timestamps to be >= the...
25658 Original commit message from CVS:
25659 * ext/ogg/gstoggmux.c: (gst_ogg_mux_clear),
25660 (gst_ogg_mux_push_buffer):
25661 Don't generate out-of-order timestamps from oggmux, instead clamp
25662 output timestamps to be >= the previously output ts.
25665 2006-09-18 10:18:22 +0000 Michael Smith <msmith@xiph.org>
25667 gst/tcp/gstmultifdsink.c: Updates, fixes, and typo corrections for multifdsink. No functional changes.
25668 Original commit message from CVS:
25669 * gst/tcp/gstmultifdsink.c: (gst_sync_method_get_type),
25670 (gst_multi_fd_sink_class_init):
25671 Updates, fixes, and typo corrections for multifdsink. No functional
25674 2006-09-17 21:58:06 +0000 Michael Smith <msmith@xiph.org>
25676 gst/typefind/gsttypefindfunctions.c: Don't crash on truncated files - check that we got an 8 byte buffer before tryin...
25677 Original commit message from CVS:
25678 * gst/typefind/gsttypefindfunctions.c: (ogganx_type_find):
25679 Don't crash on truncated files - check that we got an 8 byte buffer
25680 before trying to memcmp it.
25682 2006-09-17 20:32:09 +0000 Tim-Philipp Müller <tim@centricular.net>
25684 gst/playback/gstplaybasebin.c: Make stream-switching appear instant to the application (ie. make sure that a g_object...
25685 Original commit message from CVS:
25686 * gst/playback/gstplaybasebin.c: (get_active_source):
25687 Make stream-switching appear instant to the application
25688 (ie. make sure that a g_object_get on 'current-foo' returns
25689 the stream previously set with g_object_set(). Totem needs
25690 this to update stream-related meta-info (like audio-codec)
25691 correctly when switching streams.
25693 2006-09-17 20:14:43 +0000 Tim-Philipp Müller <tim@centricular.net>
25695 ext/alsa/gstalsamixer.c: Try harder to guess which mixer track is the master mixer track (instead of just taking the ...
25696 Original commit message from CVS:
25697 * ext/alsa/gstalsamixer.c: (gst_alsa_mixer_find_master_mixer),
25698 (gst_alsa_mixer_ensure_track_list):
25699 Try harder to guess which mixer track is the master mixer
25700 track (instead of just taking the first one that has a pvolume).
25703 2006-09-17 11:24:21 +0000 Stefan Kost <ensonic@users.sourceforge.net>
25705 gst/audioconvert/gstaudioconvert.c: Get structure-name just once.
25706 Original commit message from CVS:
25707 * gst/audioconvert/gstaudioconvert.c: (set_structure_widths),
25708 (gst_audio_convert_transform_caps):
25709 Get structure-name just once.
25711 2006-09-16 22:30:50 +0000 Stefan Kost <ensonic@users.sourceforge.net>
25713 tests/check/: Fix big batch of compiler warnings.
25714 Original commit message from CVS:
25715 * tests/check/elements/audioresample.c: (GST_START_TEST):
25716 * tests/check/elements/videotestsrc.c: (check_rgb_buf):
25717 * tests/check/elements/volume.c: (GST_START_TEST):
25718 * tests/check/elements/vorbisdec.c: (GST_START_TEST):
25719 * tests/check/pipelines/oggmux.c: (validate_ogg_page), (eos_watch),
25720 (test_pipeline), (GST_START_TEST):
25721 * tests/check/pipelines/theoraenc.c: (GST_START_TEST):
25722 * tests/check/pipelines/vorbisenc.c: (GST_START_TEST):
25723 Fix big batch of compiler warnings.
25725 2006-09-16 21:54:48 +0000 Stefan Kost <ensonic@users.sourceforge.net>
25727 ext/gnomevfs/gstgnomevfssrc.c: Add docs about icydemux usage in connection with gnomevfssrc
25728 Original commit message from CVS:
25729 * ext/gnomevfs/gstgnomevfssrc.c:
25730 Add docs about icydemux usage in connection with gnomevfssrc
25731 * ext/libvisual/visual.c:
25732 * ext/ogg/gstoggaviparse.c:
25733 * ext/ogg/gstoggdemux.c:
25734 * ext/ogg/gstoggmux.c:
25735 * ext/ogg/gstoggparse.c:
25736 * gst-libs/gst/audio/gstaudiofiltertemplate.c:
25737 * gst-libs/gst/audio/gstaudiosink.c:
25738 * gst-libs/gst/audio/gstaudiosrc.c:
25739 * gst/audiorate/gstaudiorate.c:
25740 More G_OBJECT macro fixing.
25741 * gst/audiotestsrc/gstaudiotestsrc.h:
25742 Fix wrong info in header due to copy & paste
25744 2006-09-15 14:53:44 +0000 Wim Taymans <wim.taymans@gmail.com>
25746 gst-libs/gst/audio/: Do the delay calculation in the source/sink base classes as this is specific for the capture/pla...
25747 Original commit message from CVS:
25748 * gst-libs/gst/audio/gstbaseaudiosink.c:
25749 (gst_base_audio_sink_get_time), (gst_base_audio_sink_callback):
25750 * gst-libs/gst/audio/gstbaseaudiosrc.c:
25751 (gst_base_audio_src_get_time), (gst_base_audio_src_fixate),
25752 (gst_base_audio_src_get_times), (gst_base_audio_src_get_offset),
25753 (gst_base_audio_src_create), (gst_base_audio_src_change_state):
25754 Do the delay calculation in the source/sink base classes as this is
25755 specific for the capture/playback mode.
25756 Try to fixate a bit better, like round depth up to a multiple of 8
25758 Handle underruns correctly by marking DISCONT on buffers and adjusting
25759 timestamps to handle the gap.
25760 Set offset/offset_end correctly on buffers.
25761 * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_pause),
25762 (gst_ring_buffer_samples_done), (gst_ring_buffer_commit),
25763 (gst_ring_buffer_read):
25764 Remove resync and underrun recovery from the ringbuffer.
25765 Fix ringbuffer read code on under/overrun.
25767 2006-09-15 11:17:02 +0000 Wim Taymans <wim.taymans@gmail.com>
25769 gst/playback/gstplaybasebin.*: Don't use a 0 low watermark when buffering, it is catching starvation way too late. In...
25770 Original commit message from CVS:
25771 * gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init),
25772 (gst_play_base_bin_init), (fill_buffer), (check_queue),
25773 (queue_threshold_reached), (gst_play_base_bin_set_property),
25774 (gst_play_base_bin_get_property):
25775 * gst/playback/gstplaybasebin.h:
25776 Don't use a 0 low watermark when buffering, it is catching starvation
25777 way too late. Instead, use a 3 second queue with 30 and 95
25778 percent low/high watermarks.
25779 Added queue-min-threshold property to configure low watermark.
25780 Use new _buffering message API.
25781 Make queue_threshold variable big enough to store a uint64 time value.
25782 API: playbin::queue-min-threshold property.
25784 2006-09-15 09:13:50 +0000 Wim Taymans <wim.taymans@gmail.com>
25786 configure.ac: We require 0.10.10.1 now because of _wait_preroll().
25787 Original commit message from CVS:
25789 We require 0.10.10.1 now because of _wait_preroll().
25790 * gst-libs/gst/audio/gstbaseaudiosink.c:
25791 (gst_base_audio_sink_render):
25792 Use gst_base_sink_wait_preroll().
25794 2006-09-15 09:09:00 +0000 Wim Taymans <wim.taymans@gmail.com>
25796 ext/alsa/: Use DEBUG_OBJECT more.
25797 Original commit message from CVS:
25798 * ext/alsa/gstalsasink.c: (xrun_recovery), (gst_alsasink_write):
25799 * ext/alsa/gstalsasrc.c: (xrun_recovery), (gst_alsasrc_read):
25800 Use DEBUG_OBJECT more.
25802 === release 0.10.10 ===
25804 2006-09-14 20:09:19 +0000 Thomas Vander Stichele <thomas@apestaart.org>
25811 * docs/plugins/gst-plugins-base-plugins.args:
25812 * docs/plugins/inspect/plugin-adder.xml:
25813 * docs/plugins/inspect/plugin-alsa.xml:
25814 * docs/plugins/inspect/plugin-audioconvert.xml:
25815 * docs/plugins/inspect/plugin-audiorate.xml:
25816 * docs/plugins/inspect/plugin-audioresample.xml:
25817 * docs/plugins/inspect/plugin-audiotestsrc.xml:
25818 * docs/plugins/inspect/plugin-cdparanoia.xml:
25819 * docs/plugins/inspect/plugin-decodebin.xml:
25820 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
25821 * docs/plugins/inspect/plugin-gdp.xml:
25822 * docs/plugins/inspect/plugin-gnomevfs.xml:
25823 * docs/plugins/inspect/plugin-libvisual.xml:
25824 * docs/plugins/inspect/plugin-ogg.xml:
25825 * docs/plugins/inspect/plugin-pango.xml:
25826 * docs/plugins/inspect/plugin-playbin.xml:
25827 * docs/plugins/inspect/plugin-subparse.xml:
25828 * docs/plugins/inspect/plugin-tcp.xml:
25829 * docs/plugins/inspect/plugin-theora.xml:
25830 * docs/plugins/inspect/plugin-typefindfunctions.xml:
25831 * docs/plugins/inspect/plugin-video4linux.xml:
25832 * docs/plugins/inspect/plugin-videorate.xml:
25833 * docs/plugins/inspect/plugin-videoscale.xml:
25834 * docs/plugins/inspect/plugin-videotestsrc.xml:
25835 * docs/plugins/inspect/plugin-volume.xml:
25836 * docs/plugins/inspect/plugin-vorbis.xml:
25837 * docs/plugins/inspect/plugin-ximagesink.xml:
25838 * docs/plugins/inspect/plugin-xvimagesink.xml:
25839 * ext/theora/theoraparse.c:
25840 * gst-libs/gst/rtp/gstrtpbuffer.c:
25841 * gst/playback/gstplaybin.c:
25842 * tests/check/Makefile.am:
25843 * win32/common/config.h:
25845 Original commit message from CVS:
25848 2006-09-09 16:08:06 +0000 Thomas Vander Stichele <thomas@apestaart.org>
25851 * win32/common/config.h:
25853 Original commit message from CVS:
25856 2006-09-07 19:01:23 +0000 Thomas Vander Stichele <thomas@apestaart.org>
25859 update bug in changelog
25860 Original commit message from CVS:
25861 update bug in changelog
25863 2006-09-07 19:00:33 +0000 Michael Smith <msmith@fluendo.com>
25865 Fix implementation of sync-method 'next-keyframe'
25866 Original commit message from CVS:
25867 patch by: Michael Smith <msmith at fluendo dot com>
25868 * gst/tcp/gstmultifdsink.c: (is_sync_frame),
25869 (gst_multi_fd_sink_client_queue_buffer),
25870 (gst_multi_fd_sink_new_client):
25871 * tests/check/elements/multifdsink.c: (GST_START_TEST),
25872 (multifdsink_suite):
25873 Fix implementation of sync-method 'next-keyframe'
25875 2006-09-07 15:00:08 +0000 Wim Taymans <wim@fluendo.com>
25877 ext/gnomevfs/gstgnomevfssrc.c: This patch removes the RANDOM flag that was incorrectly introduced with revision 1.91....
25878 Original commit message from CVS:
25879 patch by: Wim Taymans <wim at fluendo dot com>
25880 * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_start):
25881 This patch removes the RANDOM flag that was incorrectly introduced with
25882 revision 1.91. Fixes #354590
25884 2006-09-07 14:56:47 +0000 Thomas Vander Stichele <thomas@apestaart.org>
25887 * win32/common/config.h:
25889 Original commit message from CVS:
25892 2006-09-07 14:56:31 +0000 Thomas Vander Stichele <thomas@apestaart.org>
25909 Original commit message from CVS:
25912 2006-09-05 09:12:25 +0000 Tim-Philipp Müller <tim@centricular.net>
25914 tests/check/Makefile.am: Random variation in Makefile line to see if it makes the gen64-base-full bot any happier.
25915 Original commit message from CVS:
25916 * tests/check/Makefile.am:
25917 Random variation in Makefile line to see if it makes the
25918 gen64-base-full bot any happier.
25920 2006-09-04 19:04:35 +0000 Tim-Philipp Müller <tim@centricular.net>
25922 tests/check/pipelines/oggmux.c: Disable test that fails at the moment (killed after timeout).
25923 Original commit message from CVS:
25924 * tests/check/pipelines/oggmux.c: (oggmux_suite):
25925 Disable test that fails at the moment (killed after timeout).
25927 2006-09-04 18:19:06 +0000 James Livingston <doclivingston@gmail.com>
25929 tests/check/: Add simple unit test for oggmux from #337026 with checking for the
25930 Original commit message from CVS:
25931 Patch by: James Livingston <doclivingston at gmail.com>
25932 * tests/check/Makefile.am:
25933 * tests/check/pipelines/.cvsignore:
25934 * tests/check/pipelines/oggmux.c: (get_page_codec),
25935 (check_chain_final_state), (fail_if_audio), (validate_ogg_page),
25936 (eos_buffer_probe), (start_pipeline), (stop_pipeline), (eos_watch),
25937 (test_pipeline), (test_vorbis), (test_theora), (test_vorbis_theora),
25938 (test_theora_vorbis), (oggmux_suite):
25939 Add simple unit test for oggmux from #337026 with checking for the
25940 EOS flags disabled for the time being.
25942 2006-09-04 09:13:01 +0000 Alessandro Dessina <alessandro@nnva.org>
25944 ext/ogg/gstoggmux.c: Add cmml caps to oggmux. Fixes #353912
25945 Original commit message from CVS:
25946 patch by: Alessandro Dessina <alessandro nnva org>
25947 * ext/ogg/gstoggmux.c:
25948 Add cmml caps to oggmux. Fixes #353912
25950 2006-09-02 13:20:59 +0000 Tim-Philipp Müller <tim@centricular.net>
25952 tests/check/elements/videotestsrc.c: Returning a return value often helps. In this case, we don't need the return val...
25953 Original commit message from CVS:
25954 * tests/check/elements/videotestsrc.c: (check_rgb_buf):
25955 Returning a return value often helps. In this case, we
25956 don't need the return value anyway, so just get rid of it.
25957 Should make build bots much happier.
25959 2006-09-02 12:59:48 +0000 Tim-Philipp Müller <tim@centricular.net>
25961 gst/videotestsrc/videotestsrc.*: Add support for AYUV and the various RGBA formats. Initialise fields of paintinfo st...
25962 Original commit message from CVS:
25963 * gst/videotestsrc/videotestsrc.c: (paintinfo_find_by_structure),
25964 (paint_get_structure), (gst_video_test_src_get_size),
25965 (gst_video_test_src_smpte), (gst_video_test_src_snow),
25966 (gst_video_test_src_unicolor), (paint_setup_AYUV),
25967 (paint_hline_AYUV), (paint_setup_ARGB8888), (paint_setup_ABGR8888),
25968 (paint_setup_RGBA8888), (paint_setup_BGRA8888), (paint_hline_str4):
25969 * gst/videotestsrc/videotestsrc.h:
25970 Add support for AYUV and the various RGBA formats. Initialise
25971 fields of paintinfo structs allocated on the stack.
25972 * tests/check/elements/videotestsrc.c: (right_shift_colour),
25973 (fix_expected_colour), (check_rgb_buf), (got_buf_cb),
25974 (GST_START_TEST), (videotestsrc_suite):
25975 Add unit tests for videotestsrc's RGB output.
25977 2006-09-01 16:12:35 +0000 Tim-Philipp Müller <tim@centricular.net>
25979 gst/videotestsrc/: Add more uni-colour patterns ("white", "red", "green", and "blue").
25980 Original commit message from CVS:
25981 * gst/videotestsrc/gstvideotestsrc.c:
25982 (gst_video_test_src_pattern_get_type),
25983 (gst_video_test_src_set_pattern):
25984 * gst/videotestsrc/gstvideotestsrc.h:
25985 * gst/videotestsrc/videotestsrc.c: (gst_video_test_src_unicolor),
25986 (gst_video_test_src_black), (gst_video_test_src_white),
25987 (gst_video_test_src_red), (gst_video_test_src_green),
25988 (gst_video_test_src_blue):
25989 * gst/videotestsrc/videotestsrc.h:
25990 Add more uni-colour patterns ("white", "red", "green", and "blue").
25992 2006-09-01 10:07:05 +0000 Tim-Philipp Müller <tim@centricular.net>
25994 gst/videotestsrc/videotestsrc.c: Fix stride for YVYU, should be word-aligned (#353658).
25995 Original commit message from CVS:
25996 * gst/videotestsrc/videotestsrc.c: (paint_setup_YVYU):
25997 Fix stride for YVYU, should be word-aligned (#353658).
25999 2006-08-31 14:37:33 +0000 Tim-Philipp Müller <tim@centricular.net>
26001 gst/adder/gstadder.c: Fix build.
26002 Original commit message from CVS:
26003 * gst/adder/gstadder.c: (gst_adder_src_event):
26006 2006-08-31 12:39:17 +0000 Edward Hervey <bilboed@bilboed.com>
26008 gst/adder/gstadder.*: Remember the start position asked in the incoming seeks, so we can output GST_EVENT_NEW_SEGMENT...
26009 Original commit message from CVS:
26010 * gst/adder/gstadder.c: (forward_event_func),
26011 (gst_adder_src_event), (gst_adder_collected),
26012 (gst_adder_change_state):
26013 * gst/adder/gstadder.h:
26014 Remember the start position asked in the incoming seeks, so we can
26015 output GST_EVENT_NEW_SEGMENT with a correct position value (instead
26016 of assuming it will always be 0).
26018 2006-08-31 12:31:00 +0000 Edward Hervey <bilboed@bilboed.com>
26020 ext/ogg/gstoggdemux.c: Send the GST_EVENT_NEW_SEGMENT from the streaming thread.
26021 Original commit message from CVS:
26022 * ext/ogg/gstoggdemux.c: (gst_ogg_demux_init),
26023 (gst_ogg_demux_finalize), (gst_ogg_demux_perform_seek),
26024 (gst_ogg_demux_loop):
26025 Send the GST_EVENT_NEW_SEGMENT from the streaming thread.
26027 2006-08-30 17:22:27 +0000 Tim-Philipp Müller <tim@centricular.net>
26029 gst/ffmpegcolorspace/gstffmpegcolorspace.c: Return FALSE instead of returning a random false unit size when the forma...
26030 Original commit message from CVS:
26031 * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
26032 (gst_ffmpegcsp_get_unit_size):
26033 Return FALSE instead of returning a random false unit
26034 size when the format isn't known/supported (even if
26035 this shouldn't happen under normal circumstances).
26037 2006-08-29 15:23:46 +0000 Tim-Philipp Müller <tim@centricular.net>
26039 ext/gnomevfs/gstgnomevfssrc.c: Try harder to get the size from a uri by using _info_uri() when _info_from_handle() do...
26040 Original commit message from CVS:
26041 Patch by: Tim-Philipp Müller <tim at centricular dot net>
26042 * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_create),
26043 (gst_gnome_vfs_src_start):
26044 Try harder to get the size from a uri by using _info_uri() when
26045 _info_from_handle() does not give us enough info.
26046 Also follow symlinks when getting the size.
26047 Partially Fixes #332864.
26049 2006-08-29 11:50:51 +0000 Viktor Peters <viktor.peters@gmail.com>
26051 ext/alsa/: Improve and fix mixer track handling, in particular better handling of alsa's pvolume/pswitch/cvolume/cswi...
26052 Original commit message from CVS:
26053 Patch by: Viktor Peters <viktor dot peters at gmail dot com>
26054 * ext/alsa/gstalsamixer.c: (gst_alsa_mixer_ensure_track_list),
26055 (gst_alsa_mixer_update), (gst_alsa_mixer_get_volume),
26056 (gst_alsa_mixer_set_volume), (gst_alsa_mixer_set_mute),
26057 (gst_alsa_mixer_set_record):
26058 * ext/alsa/gstalsamixertrack.c:
26059 (gst_alsa_mixer_track_update_alsa_capabilities),
26060 (alsa_track_has_cap), (gst_alsa_mixer_track_new),
26061 (gst_alsa_mixer_track_update):
26062 * ext/alsa/gstalsamixertrack.h:
26063 Improve and fix mixer track handling, in particular better handling
26064 of alsa's pvolume/pswitch/cvolume/cswitch capabilities; create separate
26065 track objects for tracks that have both capture and playback volume
26066 (and label them differently as well so they're not mistakenly
26067 assumed to be duplicates); classify mixer tracks that only affect
26068 the audible volume of something (rather than the capture volume)
26069 as playback tracks. Redefine/fix meaning of RECORD and MUTE flags
26070 for capture tracks to correspond to alsa-pswitch alsa-cswitch
26071 (following the meaning documented in the mixer interface header
26072 file); add support for alsa's exclusive cswitch groups; update/sync
26073 state/flags better if mixer settings are changed by another
26074 application. Fixes #336075.
26076 2006-08-29 10:58:43 +0000 Tim-Philipp Müller <tim@centricular.net>
26078 gst/playback/gstplaybin.c: Improve docs: add section about BUFFERING messages sent by playbin.
26079 Original commit message from CVS:
26080 * gst/playback/gstplaybin.c:
26081 Improve docs: add section about BUFFERING messages sent by playbin.
26083 2006-08-29 10:51:12 +0000 Michael Smith <msmith@xiph.org>
26085 ext/vorbis/vorbisenc.c: Ignore explicit DISCONT marked on buffers (which is often spurious, particularly when using m...
26086 Original commit message from CVS:
26087 * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_push_buffer),
26088 (gst_vorbis_enc_buffer_check_discontinuous),
26089 (gst_vorbis_enc_chain):
26090 Ignore explicit DISCONT marked on buffers (which is often spurious,
26091 particularly when using multiple segments), in favour of solely
26092 using the timestamps/durations.
26094 2006-08-29 10:32:34 +0000 Edward Hervey <bilboed@bilboed.com>
26096 gst/audiorate/gstaudiorate.c: Don't rely on incoming buffers offset anymore, since it is completely broken when using...
26097 Original commit message from CVS:
26098 * gst/audiorate/gstaudiorate.c: (gst_audio_rate_chain):
26099 Don't rely on incoming buffers offset anymore, since it is completely
26100 broken when using multiple segments.
26101 Instead convert the incoming buffers timestamp to running time, and
26102 then convert that value to the offsets.
26103 Also inform GstSegment of the last outputted stop position, which is
26104 needed if we received several segments with an unknown stop value.
26106 2006-08-29 08:03:05 +0000 Thomas Vander Stichele <thomas@apestaart.org>
26108 ext/vorbis/vorbisenc.c: fix buffer unreffing on a header push failure
26109 Original commit message from CVS:
26110 * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_chain):
26111 fix buffer unreffing on a header push failure
26113 2006-08-28 16:17:13 +0000 Wim Taymans <wim.taymans@gmail.com>
26115 gst/audiorate/gstaudiorate.c: Make the metadata of the buffer writable before changing its flags.
26116 Original commit message from CVS:
26117 * gst/audiorate/gstaudiorate.c: (gst_audio_rate_sink_event),
26118 (gst_audio_rate_chain):
26119 Make the metadata of the buffer writable before changing its
26122 2006-08-28 16:09:57 +0000 Wim Taymans <wim.taymans@gmail.com>
26125 Fix changelog with bugzilla bug it fixed.
26126 Original commit message from CVS:
26127 Fix changelog with bugzilla bug it fixed.
26129 2006-08-28 16:08:18 +0000 Wim Taymans <wim.taymans@gmail.com>
26131 gst/audiorate/gstaudiorate.c: Fix audiorate some more.
26132 Original commit message from CVS:
26133 * gst/audiorate/gstaudiorate.c: (gst_audio_rate_reset),
26134 (gst_audio_rate_setcaps), (gst_audio_rate_init),
26135 (gst_audio_rate_sink_event), (gst_audio_rate_src_event),
26136 (gst_audio_rate_chain), (gst_audio_rate_change_state):
26137 Fix audiorate some more.
26138 Reset and resync counters on flush and READY.
26139 Handle the DISCONT flag correctly.
26140 Use GstSegment to track position.
26141 Fail when not negotiated.
26143 2006-08-25 16:48:28 +0000 Michael Smith <msmith@xiph.org>
26145 gst/tcp/gstmultifdsink.c: Fix spelling.
26146 Original commit message from CVS:
26147 * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_render):
26149 Remove accidently included debug line.
26151 2006-08-25 16:39:38 +0000 Wim Taymans <wim.taymans@gmail.com>
26153 gst/tcp/gstmultifdsink.c: Small cleanups.
26154 Original commit message from CVS:
26155 * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_render):
26157 If a buffer is received with no caps, make the buffer metadata
26158 writable and set the caps, making sure that we don't screw up the
26161 2006-08-25 16:19:55 +0000 Michael Smith <msmith@xiph.org>
26163 gst/gdp/gstgdppay.c: Fix memory leaks and misleading debug messages, add a couple of comments.
26164 Original commit message from CVS:
26165 * gst/gdp/gstgdppay.c: (gst_gdp_pay_reset),
26166 (gst_gdp_pay_reset_streamheader), (gst_gdp_pay_chain):
26167 Fix memory leaks and misleading debug messages, add a couple of
26169 * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_get_stats),
26170 (gst_multi_fd_sink_render):
26171 Do not use gst_buffer_make_writable() in a basesink render method,
26172 as it may incorrectly unref the buffer. Instead, use convoluted
26173 dance to avoid copying the buffer except when we need to.
26175 2006-08-25 09:54:56 +0000 Michael Smith <msmith@xiph.org>
26177 ext/vorbis/vorbisenc.c: Allow very small discontinuities in the timestamps. These we can't do anything useful with an...
26178 Original commit message from CVS:
26179 * ext/vorbis/vorbisenc.c:
26180 (gst_vorbis_enc_buffer_check_discontinuous):
26181 Allow very small discontinuities in the timestamps. These we can't
26182 do anything useful with anyway (because vorbis's timestamps have
26183 only sample granularity), and are commonly produced by elements with
26184 minor bugs. Allow up to 1/2 a sample out.
26187 2006-08-24 11:18:56 +0000 Wim Taymans <wim.taymans@gmail.com>
26189 tests/examples/seek/seek.c: Add a checkbox to enable play scrubbing. Makes it possible to disable normal scrubbing.
26190 Original commit message from CVS:
26191 * tests/examples/seek/seek.c: (seek_cb), (start_seek), (stop_seek),
26192 (play_scrub_toggle_cb), (main):
26193 Add a checkbox to enable play scrubbing. Makes it possible to disable
26196 2006-08-23 19:37:50 +0000 Stefan Kost <ensonic@users.sourceforge.net>
26198 tests/check/elements/.cvsignore: make buildbot happy
26199 Original commit message from CVS:
26200 * tests/check/elements/.cvsignore:
26201 make buildbot happy
26203 2006-08-23 16:43:03 +0000 Tim-Philipp Müller <tim@centricular.net>
26205 ext/ogg/gstogmparse.c: Refactor ogm parse, do better input checking, misc. clean-ups.
26206 Original commit message from CVS:
26207 * ext/ogg/gstogmparse.c: (gst_ogm_audio_parse_base_init),
26208 (gst_ogm_video_parse_base_init), (gst_ogm_text_parse_base_init),
26209 (gst_ogm_parse_class_init), (gst_ogm_parse_dispose),
26210 (gst_ogm_parse_init), (gst_ogm_audio_parse_init),
26211 (gst_ogm_video_parse_init), (gst_ogm_text_parse_init),
26212 (gst_ogm_parse_stream_header), (gst_ogm_parse_comment_packet),
26213 (gst_ogm_text_parse_strip_trailing_zeroes),
26214 (gst_ogm_parse_data_packet), (gst_ogm_parse_chain),
26215 (gst_ogm_parse_sink_event), (gst_ogm_parse_change_state):
26216 Refactor ogm parse, do better input checking, misc. clean-ups.
26217 Cache incoming events and push them once the source pad has
26218 been created. Don't pass unterminated strings to sscanf().
26219 Strip trailing zeroes from subtitle text output, since they
26220 are not valid UTF-8. Don't push vorbiscomment packets on
26221 the subtitle text pad. Output perfect streams if possible.
26223 2006-08-23 15:27:38 +0000 Wim Taymans <wim.taymans@gmail.com>
26225 tests/check/libs/cddabasesrc.c: Waits for tasks to settle down so that we clean up correctly for valgrind.
26226 Original commit message from CVS:
26227 * tests/check/libs/cddabasesrc.c: (GST_START_TEST):
26228 Waits for tasks to settle down so that we clean up correctly for
26231 2006-08-23 15:11:56 +0000 Tim-Philipp Müller <tim@centricular.net>
26233 tests/check/libs/tag.c: Unit test fixes: \377 is more likely to fit into 8 bits than \777; actually return return val...
26234 Original commit message from CVS:
26235 * tests/check/libs/tag.c: (GST_START_TEST), (taglists_are_equal):
26236 Unit test fixes: \377 is more likely to fit into 8 bits than \777;
26237 actually return return value in taglists_are_equal.
26239 2006-08-23 12:14:20 +0000 Tim-Philipp Müller <tim@centricular.net>
26241 ext/ogg/gstogmparse.c: Fix crash due to broken bitstream parsing on x86-64: can't make any assumptions about sizeof(s...
26242 Original commit message from CVS:
26243 * ext/ogg/gstogmparse.c: (gst_ogm_parse_chain):
26244 Fix crash due to broken bitstream parsing on x86-64: can't make
26245 any assumptions about sizeof(struct) due to alignment/packing
26246 differences on different architectures. Fixes #351790.
26248 2006-08-22 16:31:47 +0000 Wim Taymans <wim.taymans@gmail.com>
26250 gst-libs/gst/riff/riff-read.c: Protect public functions against bad input.
26251 Original commit message from CVS:
26252 * gst-libs/gst/riff/riff-read.c: (gst_riff_read_chunk),
26253 (gst_riff_parse_chunk), (gst_riff_parse_file_header),
26254 (gst_riff_parse_strh), (gst_riff_parse_strf_vids),
26255 (gst_riff_parse_strf_auds), (gst_riff_parse_strf_iavs),
26256 (gst_riff_parse_info):
26257 Protect public functions against bad input.
26261 2006-08-22 15:50:36 +0000 Tim-Philipp Müller <tim@centricular.net>
26263 gst-libs/gst/riff/: Add voxware audio IDs (even if we can't play it) (#351795).
26264 Original commit message from CVS:
26265 * gst-libs/gst/riff/riff-ids.h:
26266 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
26267 Add voxware audio IDs (even if we can't play it) (#351795).
26269 2006-08-22 15:11:52 +0000 Tim-Philipp Müller <tim@centricular.net>
26271 gst-libs/gst/riff/riff-media.c: Const-ify some arrays and use G_N_ELEMENTS instead of wasting oodles of RAM on termin...
26272 Original commit message from CVS:
26273 * gst-libs/gst/riff/riff-media.c:
26274 (gst_riff_create_video_template_caps),
26275 (gst_riff_create_audio_template_caps),
26276 (gst_riff_create_iavs_template_caps):
26277 Const-ify some arrays and use G_N_ELEMENTS instead
26278 of wasting oodles of RAM on terminator bits.
26280 2006-08-22 08:27:07 +0000 Tim-Philipp Müller <tim@centricular.net>
26282 And the same for _to_vorbiscomment_buffer(): allow id_data_len == 0 for speex.
26283 Original commit message from CVS:
26284 * gst-libs/gst/tag/gstvorbistag.c:
26285 (gst_tag_list_to_vorbiscomment_buffer):
26286 * tests/check/libs/tag.c: (GST_START_TEST):
26287 And the same for _to_vorbiscomment_buffer(): allow
26288 id_data_len == 0 for speex.
26290 2006-08-21 19:04:14 +0000 Thomas Vander Stichele <thomas@apestaart.org>
26294 Original commit message from CVS:
26297 2006-08-21 19:01:41 +0000 Thomas Vander Stichele <thomas@apestaart.org>
26299 Move GDP plugin to -base from -bad. Closes #347783.
26300 Original commit message from CVS:
26302 * docs/plugins/Makefile.am:
26303 * docs/plugins/gst-plugins-base-plugins-docs.sgml:
26304 * docs/plugins/gst-plugins-base-plugins-sections.txt:
26305 * docs/plugins/inspect/plugin-gdp.xml:
26306 * gst/gdp/Makefile.am:
26307 * tests/check/Makefile.am:
26308 Move GDP plugin to -base from -bad. Closes #347783.
26310 2006-08-21 18:34:46 +0000 Tim-Philipp Müller <tim@centricular.net>
26312 gst-libs/gst/tag/gstvorbistag.c: Allow id_data_len == 0 (needed for vorbis comments in Speex files).
26313 Original commit message from CVS:
26314 * gst-libs/gst/tag/gstvorbistag.c:
26315 (gst_tag_list_from_vorbiscomment_buffer):
26316 Allow id_data_len == 0 (needed for vorbis comments in Speex files).
26317 Also add some checks to make sure we don't memcmp() beyond the end of
26318 vorbiscomment buffer if the ID to check for is larger than the buffer.
26319 * tests/check/libs/tag.c: (GST_START_TEST):
26320 Some more tests for gst_tag_list_from_vorbiscomment_buffer().
26322 2006-08-21 16:39:25 +0000 Tim-Philipp Müller <tim@centricular.net>
26324 ext/vorbis/vorbisenc.c: Use vorbis comment utility functions from libgsttag instead of re-inventing the wheel (partia...
26325 Original commit message from CVS:
26326 * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_metadata_set1),
26327 (gst_vorbis_enc_set_metadata):
26328 Use vorbis comment utility functions from libgsttag
26329 instead of re-inventing the wheel (partially fixes #347091).
26331 2006-08-21 11:42:12 +0000 Jan Schmidt <thaytan@mad.scientist.com>
26333 tests/check/elements/audioconvert.c: Fix leaks. Wait for state transitions that might happen ASYNC, as well as some t...
26334 Original commit message from CVS:
26335 * tests/check/elements/audioconvert.c: (GST_START_TEST):
26336 Fix leaks. Wait for state transitions that might happen ASYNC, as well
26337 as some that won't.
26339 2006-08-21 10:32:51 +0000 Wim Taymans <wim.taymans@gmail.com>
26341 docs/libs/: Don't try to GObject scan the netbuffer as it's not a GObject.
26342 Original commit message from CVS:
26343 * docs/libs/Makefile.am:
26344 * docs/libs/gst-plugins-base-libs-sections.txt:
26345 * docs/libs/gst-plugins-base-libs.types:
26346 Don't try to GObject scan the netbuffer as it's not a GObject.
26348 * gst-libs/gst/netbuffer/gstnetbuffer.c:
26349 * gst-libs/gst/netbuffer/gstnetbuffer.h:
26350 Document GstNetBuffer.
26352 2006-08-21 08:54:06 +0000 Stefan Kost <ensonic@users.sourceforge.net>
26354 tests/check/elements/audioconvert.c: Add testcase for caps-size-explosion
26355 Original commit message from CVS:
26356 * tests/check/elements/audioconvert.c: (GST_START_TEST),
26357 (audioconvert_suite):
26358 Add testcase for caps-size-explosion
26360 2006-08-20 13:05:44 +0000 Stefan Kost <ensonic@users.sourceforge.net>
26362 gst/audioconvert/gstaudioconvert.c: Lower debug, use g_assert in _get_unit_size
26363 Original commit message from CVS:
26364 * gst/audioconvert/gstaudioconvert.c:
26365 (gst_audio_convert_get_unit_size), (set_structure_widths):
26366 Lower debug, use g_assert in _get_unit_size
26367 * gst/audioresample/gstaudioresample.c:
26368 (audioresample_get_unit_size):
26369 * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
26370 (gst_ffmpegcsp_get_unit_size):
26371 * gst/videoscale/gstvideoscale.c: (gst_video_scale_get_unit_size):
26372 use g_assert in _get_unit_size
26374 2006-08-18 21:21:48 +0000 Tim-Philipp Müller <tim@centricular.net>
26377 ChangeLog surgery: fix bug number
26378 Original commit message from CVS:
26379 ChangeLog surgery: fix bug number
26381 2006-08-18 16:43:26 +0000 Wim Taymans <wim.taymans@gmail.com>
26383 Document GstRTPBuffer.
26384 Original commit message from CVS:
26385 * docs/libs/gst-plugins-base-libs-sections.txt:
26386 * gst-libs/gst/rtp/gstrtpbuffer.c: (gst_rtp_buffer_set_packet_len),
26387 (gst_rtp_buffer_pad_to), (gst_rtp_buffer_get_payload_subbuffer),
26388 (gst_rtp_buffer_get_payload_buffer):
26389 * gst-libs/gst/rtp/gstrtpbuffer.h:
26390 Document GstRTPBuffer.
26391 Added function to efficiently strip payload headers.
26392 API: gst_rtp_buffer_get_payload_subbuffer()
26394 2006-08-17 16:52:06 +0000 Tim-Philipp Müller <tim@centricular.net>
26396 gst-libs/gst/tag/gstvorbistag.c: Serialise unknown vorbis comments into GST_TAG_EXTENDED_COMMENT tags and deserialise...
26397 Original commit message from CVS:
26398 * gst-libs/gst/tag/gstvorbistag.c: (gst_vorbis_tag_add),
26399 (gst_tag_to_vorbis_comments):
26400 Serialise unknown vorbis comments into GST_TAG_EXTENDED_COMMENT
26401 tags and deserialise them properly as well (#351768).
26402 Add some more gtk-doc blurbs and also some g_return_if_fail().
26403 * tests/check/libs/tag.c: (GST_START_TEST),
26404 (back_to_vorbis_comments), (taglists_are_equal), (tag_suite):
26407 2006-08-17 15:43:40 +0000 Wim Taymans <wim.taymans@gmail.com>
26409 ext/ogg/: Added ogg-in-avi parser element. Fixes #140139.
26410 Original commit message from CVS:
26411 * ext/ogg/Makefile.am:
26412 * ext/ogg/gstogg.c: (plugin_init):
26413 * ext/ogg/gstoggaviparse.c: (gst_ogg_avi_parse_get_type),
26414 (gst_ogg_avi_parse_base_init), (gst_ogg_avi_parse_class_init),
26415 (gst_ogg_avi_parse_init), (gst_ogg_avi_parse_finalize),
26416 (gst_ogg_avi_parse_setcaps), (gst_ogg_avi_parse_event),
26417 (gst_ogg_avi_parse_push_packet), (gst_ogg_avi_parse_chain),
26418 (gst_ogg_avi_parse_change_state), (gst_ogg_avi_parse_plugin_init):
26419 Added ogg-in-avi parser element. Fixes #140139.
26420 * ext/ogg/gstoggmux.c: (gst_ogg_mux_buffer_from_page):
26421 Fixed a bug in oggdemux debug code.
26422 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps),
26423 (gst_riff_create_audio_template_caps):
26424 Recognise Ogg in the AVI extensible wave format.
26426 2006-08-17 10:00:00 +0000 Tim-Philipp Müller <tim@centricular.net>
26428 gst-libs/gst/cdda/gstcddabasesrc.c: Make buffer durations add up (duration should be next_ts-ts for perfect streams)....
26429 Original commit message from CVS:
26430 * gst-libs/gst/cdda/gstcddabasesrc.c: (gst_cdda_base_src_create):
26431 Make buffer durations add up (duration should be next_ts-ts for
26432 perfect streams). Fixes CD ripping to Ogg/Vorbis with vorbisenc
26434 * tests/check/libs/cddabasesrc.c: (gst_cd_foo_src_close),
26435 (test_buffer_timestamps), (cddabasesrc_suite):
26436 Add unit test for the above.
26437 * tests/check/Makefile.am:
26438 Don't know why cddabasesrc test was in VALGRIND_TO_FIX, remove
26439 to see what happens.
26441 2006-08-16 11:38:52 +0000 Wim Taymans <wim.taymans@gmail.com>
26443 ext/alsa/: Avoid setting and using a NULL device name.
26444 Original commit message from CVS:
26445 * ext/alsa/gstalsasink.c: (gst_alsasink_set_property),
26446 (gst_alsasink_open):
26447 * ext/alsa/gstalsasrc.c: (gst_alsasrc_set_property),
26448 (gst_alsasrc_open):
26449 Avoid setting and using a NULL device name.
26450 Print more info when we fail to open a device.
26452 2006-08-16 11:28:57 +0000 Tim-Philipp Müller <tim@centricular.net>
26454 API: add gst_tag_parse_extended_comment() (#351426).
26455 Original commit message from CVS:
26456 * docs/libs/gst-plugins-base-libs-sections.txt:
26457 * gst-libs/gst/tag/tag.h:
26458 * gst-libs/gst/tag/tags.c: (gst_tag_parse_extended_comment):
26459 API: add gst_tag_parse_extended_comment() (#351426).
26460 * tests/check/Makefile.am:
26461 * tests/check/libs/.cvsignore:
26462 * tests/check/libs/tag.c: (GST_START_TEST), (tag_suite), (main):
26463 Add unit test for gst_tag_parse_extended_comment().
26465 2006-08-15 19:20:16 +0000 Tim-Philipp Müller <tim@centricular.net>
26467 sys/: Fix leak (#351502).
26468 Original commit message from CVS:
26469 * sys/ximage/ximagesink.c: (gst_ximagesink_get_property):
26470 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_property):
26471 Fix leak (#351502).
26473 2006-08-15 17:21:33 +0000 Tim-Philipp Müller <tim@centricular.net>
26476 Original commit message from CVS:
26477 * docs/plugins/gst-plugins-base-plugins-docs.sgml:
26478 * docs/plugins/gst-plugins-base-plugins-sections.txt:
26479 * docs/plugins/gst-plugins-base-plugins.args:
26480 * gst/playback/gstplaybin.c:
26482 * docs/plugins/inspect/plugin-adder.xml:
26483 * docs/plugins/inspect/plugin-alsa.xml:
26484 * docs/plugins/inspect/plugin-audioconvert.xml:
26485 * docs/plugins/inspect/plugin-audiorate.xml:
26486 * docs/plugins/inspect/plugin-audioresample.xml:
26487 * docs/plugins/inspect/plugin-audiotestsrc.xml:
26488 * docs/plugins/inspect/plugin-cdparanoia.xml:
26489 * docs/plugins/inspect/plugin-decodebin.xml:
26490 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
26491 * docs/plugins/inspect/plugin-gnomevfs.xml:
26492 * docs/plugins/inspect/plugin-ogg.xml:
26493 * docs/plugins/inspect/plugin-pango.xml:
26494 * docs/plugins/inspect/plugin-playbin.xml:
26495 * docs/plugins/inspect/plugin-subparse.xml:
26496 * docs/plugins/inspect/plugin-tcp.xml:
26497 * docs/plugins/inspect/plugin-theora.xml:
26498 * docs/plugins/inspect/plugin-typefindfunctions.xml:
26499 * docs/plugins/inspect/plugin-video4linux.xml:
26500 * docs/plugins/inspect/plugin-videorate.xml:
26501 * docs/plugins/inspect/plugin-videoscale.xml:
26502 * docs/plugins/inspect/plugin-videotestsrc.xml:
26503 * docs/plugins/inspect/plugin-volume.xml:
26504 * docs/plugins/inspect/plugin-vorbis.xml:
26505 * docs/plugins/inspect/plugin-ximagesink.xml:
26506 * docs/plugins/inspect/plugin-xvimagesink.xml:
26507 Update to CVS version.
26509 2006-08-14 17:54:01 +0000 Tim-Philipp Müller <tim@centricular.net>
26511 gst/playback/gstplaybin.c: Add "connection-speed" property; re-order redirect messages with multiple redirect locatio...
26512 Original commit message from CVS:
26513 * gst/playback/gstplaybin.c: (gst_play_bin_class_init),
26514 (gst_play_bin_set_property), (gst_play_bin_get_property),
26515 (value_list_append_structure_list),
26516 (gst_play_bin_handle_redirect_message),
26517 (gst_play_bin_handle_message):
26518 Add "connection-speed" property; re-order redirect messages with
26519 multiple redirect locations depending on the minimum bitrate if
26520 that information is available and a connection speed is set
26523 2006-08-14 11:41:04 +0000 Tim-Philipp Müller <tim@centricular.net>
26525 gst/playback/gstplaybin.c: Update max volume to the same value that the volume element uses.
26526 Original commit message from CVS:
26527 * gst/playback/gstplaybin.c:
26528 Update max volume to the same value that the volume element uses.
26530 2006-08-14 10:50:15 +0000 Wim Taymans <wim.taymans@gmail.com>
26532 ext/alsa/gstalsamixer.c: Less uglyness..
26533 Original commit message from CVS:
26534 * ext/alsa/gstalsamixer.c: (gst_alsa_mixer_open):
26537 2006-08-14 10:49:10 +0000 Wim Taymans <wim.taymans@gmail.com>
26539 ext/ogg/gstoggdemux.c: Add some more debug info.
26540 Original commit message from CVS:
26541 * ext/ogg/gstoggdemux.c: (gst_ogg_demux_submit_buffer),
26542 (gst_ogg_demux_get_next_page), (gst_ogg_demux_perform_seek),
26543 (gst_ogg_demux_read_chain), (gst_ogg_demux_loop):
26544 Add some more debug info.
26545 Don't crash when a seek failed.
26546 Actually return the result of the seek instead of TRUE.
26547 Ignore multiple BOS pages with the same serial so that we don't create
26548 the same stream multiple times.
26549 Post an error when we fail to do the initial seek.
26551 2006-08-13 14:34:48 +0000 Wim Taymans <wim.taymans@gmail.com>
26553 ext/alsa/gstalsa.c: Small code cleanup.
26554 Original commit message from CVS:
26555 * ext/alsa/gstalsa.c: (gst_alsa_detect_rates),
26556 (gst_alsa_detect_channels), (gst_alsa_probe_supported_formats):
26557 Small code cleanup.
26558 * ext/alsa/gstalsamixer.c: (gst_alsa_mixer_open),
26559 (gst_alsa_mixer_new):
26560 Remove hack that always set the device to hw:0*.
26561 Properly find the card name for whatever device was configured.
26562 Do some better debugging.
26564 * ext/alsa/gstalsamixerelement.c:
26565 (gst_alsa_mixer_element_set_property),
26566 (gst_alsa_mixer_element_change_state):
26568 Handle setting of a NULL device name better.
26570 2006-08-11 15:53:43 +0000 Wim Taymans <wim.taymans@gmail.com>
26572 gst/adder/gstadder.c: Don't clip float values. Fixes #350900.
26573 Original commit message from CVS:
26574 * gst/adder/gstadder.c:
26575 Don't clip float values. Fixes #350900.
26577 2006-08-11 15:33:17 +0000 Andy Wingo <wingo@pobox.com>
26579 gst/tcp/gsttcp.c: Really fix the build?
26580 Original commit message from CVS:
26581 2006-08-11 Andy Wingo <wingo@pobox.com>
26582 * gst/tcp/gsttcp.c: Really fix the build?
26584 2006-08-11 15:29:56 +0000 Andy Wingo <wingo@pobox.com>
26586 gst/tcp/gsttcp.h: For now, always disable deprecation here -- fixes the build.
26587 Original commit message from CVS:
26588 2006-08-11 Andy Wingo <wingo@pobox.com>
26589 * gst/tcp/gsttcp.h: For now, always disable deprecation here --
26592 2006-08-10 13:01:31 +0000 Tim-Philipp Müller <tim@centricular.net>
26594 gst/audioconvert/gstaudioconvert.c: Float caps shouldn't have a "signed" field.
26595 Original commit message from CVS:
26596 * gst/audioconvert/gstaudioconvert.c: (make_lossless_changes):
26597 Float caps shouldn't have a "signed" field.
26599 2006-08-10 08:56:22 +0000 Tim-Philipp Müller <tim@centricular.net>
26601 ext/ogg/gstoggdemux.c: Implement SEEKING query in its most basic form, so that we can at least check if we're seekabl...
26602 Original commit message from CVS:
26603 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_src_query):
26604 Implement SEEKING query in its most basic form, so that we can
26605 at least check if we're seekable or not (#350655).
26607 2006-08-09 14:42:58 +0000 Tim-Philipp Müller <tim@centricular.net>
26609 gst/typefind/gsttypefindfunctions.c: The checks here are not even close to anything that would justify MAXIMUM probab...
26610 Original commit message from CVS:
26611 * gst/typefind/gsttypefindfunctions.c: (mpeg2_sys_type_find):
26612 The checks here are not even close to anything that would
26613 justify MAXIMUM probability, lowering to POSSIBLE until someone
26614 fixes the checks (case at hand: quicktime redirection files
26615 might start with 00 00 01 XX and pass the checks here just
26616 fine, see #350399).
26618 2006-08-08 13:57:29 +0000 Edward Hervey <bilboed@bilboed.com>
26620 tests/check/elements/gdpdepay.c: I forgot to include the file containing the #define :)
26621 Original commit message from CVS:
26622 * tests/check/elements/gdpdepay.c: (gdpdepay_suite):
26623 I forgot to include the file containing the #define :)
26624 Now includes "config.h"
26626 2006-08-08 13:45:44 +0000 Edward Hervey <bilboed@bilboed.com>
26628 tests/check/elements/gdpdepay.c: Ignore test known to fail on PPC64. See #348114.
26629 Original commit message from CVS:
26630 * tests/check/elements/gdpdepay.c: (gdpdepay_suite):
26631 Ignore test known to fail on PPC64. See #348114.
26633 2006-08-08 08:41:13 +0000 Sjoerd Simons <sjoerd@luon.net>
26635 gst/typefind/gsttypefindfunctions.c: Better detection for multipart/x-mixed-replace: accept leading whitespaces befor...
26636 Original commit message from CVS:
26637 Patch by: Sjoerd Simons <sjoerd at luon net>
26638 * gst/typefind/gsttypefindfunctions.c: (multipart_type_find):
26639 Better detection for multipart/x-mixed-replace: accept leading
26640 whitespaces before the boundary marker as well (as our very own
26641 multipartmux used to produce) (#349068).
26643 2006-08-07 08:26:03 +0000 Young-Ho Cha <ganadist@chollian.net>
26645 gst-libs/gst/riff/: Detect DTS audio streams (#350157).
26646 Original commit message from CVS:
26647 Patch by: Young-Ho Cha <ganadist at chollian net>
26648 * gst-libs/gst/riff/riff-ids.h:
26649 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps),
26650 (gst_riff_create_audio_template_caps):
26651 Detect DTS audio streams (#350157).
26653 2006-08-05 17:08:05 +0000 Andy Wingo <wingo@pobox.com>
26655 ext/theora/gsttheoraparse.h: ext/theora/theoraparse.c (gst_theora_parse_class_init) (theora_parse_dispose, theora_par...
26656 Original commit message from CVS:
26657 2006-08-05 Andy Wingo <wingo@pobox.com>
26658 * ext/theora/gsttheoraparse.h:
26659 * ext/theora/theoraparse.c (gst_theora_parse_class_init)
26660 (theora_parse_dispose, theora_parse_set_property)
26661 (theora_parse_get_property, theora_parse_munge_granulepos)
26662 (theora_parse_push_buffer, theora_parse_change_state): Add a
26663 property 'synchronization-points' to fix badly synchronized oggs.
26665 2006-08-04 13:20:23 +0000 Zaheer Abbas Merali <zaheerabbas@merali.org>
26667 gst/gdp/gstgdpdepay.c: Fix event parsing by gdpdepay. Fixes #349916.
26668 Original commit message from CVS:
26669 2006-08-04 Zaheer Abbas Merali <zaheerabbas at merali dot org>
26670 * gst/gdp/gstgdpdepay.c: (gst_gdp_depay_chain):
26671 Fix event parsing by gdpdepay. Fixes #349916.
26673 2006-08-03 15:04:42 +0000 Tim-Philipp Müller <tim@centricular.net>
26675 tests/check/: Add a few tests for the channel position stuff in libgstaudio.
26676 Original commit message from CVS:
26677 * tests/check/Makefile.am:
26678 * tests/check/libs/.cvsignore:
26679 * tests/check/libs/audio.c: (structure_contains_channel_positions),
26680 (fixed_caps_have_channel_positions), (GST_START_TEST),
26681 (audio_suite), (main):
26682 Add a few tests for the channel position stuff in libgstaudio.
26684 2006-08-03 14:16:06 +0000 Tim-Philipp Müller <tim@centricular.net>
26686 ext/alsa/: Add support for cards that (only) do more than 8 channels, like the Delta 44 (#345188).
26687 Original commit message from CVS:
26688 * ext/alsa/gstalsa.c: (caps_add_channel_configuration),
26689 (gst_alsa_detect_channels):
26690 * ext/alsa/gstalsasink.c:
26691 Add support for cards that (only) do more than 8 channels,
26692 like the Delta 44 (#345188).
26693 * gst-libs/gst/audio/multichannel.c:
26694 (gst_audio_check_channel_positions):
26695 * gst-libs/gst/audio/multichannel.h:
26696 API: add GST_AUDIO_CHANNEL_POSITION_NONE, which stands for an
26697 unspecified channel position and cannot be combined with any
26698 of the other audio channel positions; adjust position layout
26699 checks accordingly (#345188).
26701 2006-08-03 11:15:53 +0000 Tim-Philipp Müller <tim@centricular.net>
26703 gst/typefind/gsttypefindfunctions.c: Recognise ancient RealAudio files (see #349779).
26704 Original commit message from CVS:
26705 * gst/typefind/gsttypefindfunctions.c: (plugin_init):
26706 Recognise ancient RealAudio files (see #349779).
26708 2006-08-03 09:01:25 +0000 Jens Granseuer <jensgr@gmx.net>
26710 gst/typefind/gsttypefindfunctions.c: Add typefinder for Interplay's MVE format (#348973).
26711 Original commit message from CVS:
26712 Patch by: Jens Granseuer <jensgr at gmx net>
26713 * gst/typefind/gsttypefindfunctions.c: (plugin_init):
26714 Add typefinder for Interplay's MVE format (#348973).
26716 2006-08-02 17:03:29 +0000 Marcel Moreaux <marcelm@luon.net>
26718 gst-libs/gst/rtp/gstbasertpdepayload.*: Handle RTP sequence number rollover.
26719 Original commit message from CVS:
26720 Patch by: Marcel Moreaux <marcelm at luon dot net>
26721 * gst-libs/gst/rtp/gstbasertpdepayload.c:
26722 (gst_base_rtp_depayload_add_to_queue):
26723 * gst-libs/gst/rtp/gstbasertpdepayload.h:
26724 Handle RTP sequence number rollover.
26725 Disable jitterbuffer by default.
26727 2006-08-02 16:56:19 +0000 Wim Taymans <wim.taymans@gmail.com>
26729 gst/gdp/gstgdpdepay.c: Disable seeking.
26730 Original commit message from CVS:
26731 * gst/gdp/gstgdpdepay.c: (gst_gdp_depay_init),
26732 (gst_gdp_depay_finalize), (gst_gdp_depay_sink_event),
26733 (gst_gdp_depay_src_event), (gst_gdp_depay_chain),
26734 (gst_gdp_depay_change_state):
26737 Clear adapter on disconts.
26738 Clear caps when going to READY instead of NULL
26739 * gst/gdp/gstgdppay.c: (gst_gdp_pay_class_init),
26740 (gst_gdp_pay_init), (gst_gdp_pay_finalize), (gst_gdp_pay_reset),
26741 (gst_gdp_buffer_from_caps), (gst_gdp_pay_buffer_from_buffer),
26742 (gst_gdp_buffer_from_event), (gst_gdp_pay_reset_streamheader),
26743 (gst_gdp_queue_buffer), (gst_gdp_pay_chain),
26744 (gst_gdp_pay_sink_event), (gst_gdp_pay_src_event),
26745 (gst_gdp_pay_change_state):
26746 * gst/gdp/gstgdppay.h:
26747 Reset payloader when going to READY.
26748 Fix leaked buffers in ->queue on push errors.
26751 Create packetizer in _init, free in _finalize.
26753 2006-07-31 08:48:36 +0000 Wim Taymans <wim.taymans@gmail.com>
26755 gst/gdp/gstgdpdepay.c: Consume all events except EOS because we generate events from the gdp payload instead. Fixes #...
26756 Original commit message from CVS:
26757 * gst/gdp/gstgdpdepay.c: (gst_gdp_depay_init),
26758 (gst_gdp_depay_sink_event), (gst_gdp_depay_chain):
26759 Consume all events except EOS because we generate events from
26760 the gdp payload instead. Fixes #349204
26762 2006-07-28 17:17:24 +0000 Jan Schmidt <thaytan@mad.scientist.com>
26764 gst/audioresample/gstaudioresample.c: Don't leak references to the incoming caps. Clean them up when stopping.
26765 Original commit message from CVS:
26766 * gst/audioresample/gstaudioresample.c: (audioresample_stop),
26767 (audioresample_set_caps):
26768 Don't leak references to the incoming caps. Clean them up when
26770 * gst/videoscale/gstvideoscale.c: (gst_video_scale_class_init),
26771 (gst_video_scale_finalize):
26772 Don't leak our temporary pixel buffer.
26773 * tests/check/Makefile.am:
26774 * tests/check/pipelines/simple-launch-lines.c: (run_pipeline),
26775 (GST_START_TEST), (simple_launch_lines_suite):
26776 Fix leaks and re-enable the test for valgrind checking.
26778 2006-07-28 16:39:31 +0000 Sjoerd Simons <sjoerd@luon.net>
26780 gst/typefind/gsttypefindfunctions.c: Add typefind function for multipart/x-mixed-replace (#348916).
26781 Original commit message from CVS:
26782 Patch by: Sjoerd Simons <sjoerd at luon net>
26783 * gst/typefind/gsttypefindfunctions.c: (multipart_type_find),
26785 Add typefind function for multipart/x-mixed-replace (#348916).
26787 2006-07-28 14:14:58 +0000 Wim Taymans <wim.taymans@gmail.com>
26789 gst/adder/gstadder.c: Fix leak in duration query.
26790 Original commit message from CVS:
26791 * gst/adder/gstadder.c: (gst_adder_setcaps),
26792 (gst_adder_query_duration):
26793 Fix leak in duration query.
26794 Reflow some docs and notes.
26796 2006-07-28 13:42:48 +0000 Michael Smith <msmith@xiph.org>
26798 tests/check/pipelines/vorbisenc.c: Enable Andy's extra vorbisenc test, now that it passes. Also fix one aspect of it.
26799 Original commit message from CVS:
26800 * tests/check/pipelines/vorbisenc.c: (GST_START_TEST),
26802 Enable Andy's extra vorbisenc test, now that it passes. Also fix one
26805 2006-07-28 12:48:21 +0000 Michael Smith <msmith@xiph.org>
26807 ext/vorbis/vorbisenc.*: Handle discontinuities in the input vorbis stream correctly, so that the output is properly t...
26808 Original commit message from CVS:
26809 * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_generate_sink_caps),
26810 (gst_vorbis_enc_sink_getcaps), (gst_vorbis_enc_buffer_from_packet),
26811 (gst_vorbis_enc_push_buffer),
26812 (gst_vorbis_enc_buffer_check_discontinuous),
26813 (gst_vorbis_enc_chain), (gst_vorbis_enc_change_state):
26814 * ext/vorbis/vorbisenc.h:
26815 Handle discontinuities in the input vorbis stream correctly,
26816 so that the output is properly timestamped (and has good granulepos
26817 values). Needs some oggmux fixes too.
26819 2006-07-27 10:52:52 +0000 Kai Vehmanen <kv2004@eca.cx>
26821 gst-libs/gst/rtp/gstbasertpdepayload.c: Don't send multiple newsegments with different formats.
26822 Original commit message from CVS:
26823 patch by: Kai Vehmanen <kv2004 eca cx>
26824 * gst-libs/gst/rtp/gstbasertpdepayload.c:
26825 (gst_base_rtp_depayload_chain),
26826 (gst_base_rtp_depayload_handle_sink_event),
26827 (gst_base_rtp_depayload_change_state):
26828 Don't send multiple newsegments with different formats.
26831 2006-07-26 15:20:56 +0000 Wim Taymans <wim.taymans@gmail.com>
26833 ext/ogg/gstoggdemux.c: Make seeking in ogg more accurate again by doing the more correct granuletime to stream time c...
26834 Original commit message from CVS:
26835 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_internal_chain),
26836 (gst_ogg_demux_do_seek), (gst_ogg_demux_read_chain):
26837 Make seeking in ogg more accurate again by doing the more correct
26838 granuletime to stream time conversion.
26840 2006-07-26 10:59:35 +0000 Thomas Vander Stichele <thomas@apestaart.org>
26842 gst/tcp/gstmultifdsink.c: debug a little more understandably do not use goto as a substitute for break, especially if...
26843 Original commit message from CVS:
26844 * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_add_full),
26845 (gst_multi_fd_sink_new_client):
26846 debug a little more understandably
26847 do not use goto as a substitute for break, especially if
26848 break is also being used
26850 2006-07-26 10:55:13 +0000 Thomas Vander Stichele <thomas@apestaart.org>
26852 * gst/tcp/gsttcp.c:
26853 move a recurring normal event to LOG, where it should be
26854 Original commit message from CVS:
26855 move a recurring normal event to LOG, where it should be
26857 2006-07-26 10:54:13 +0000 Thomas Vander Stichele <thomas@apestaart.org>
26859 * ext/vorbis/vorbisdec.c:
26861 Original commit message from CVS:
26864 2006-07-26 10:52:05 +0000 Thomas Vander Stichele <thomas@apestaart.org>
26866 gst/gdp/gstgdpdepay.c: proxying get/set caps is the wrong thing to do, since we really do change caps quite fundament...
26867 Original commit message from CVS:
26868 * gst/gdp/gstgdpdepay.c: (gst_gdp_depay_init):
26869 proxying get/set caps is the wrong thing to do, since we really
26870 do change caps quite fundamentally
26871 * tests/check/elements/gdpdepay.c:
26872 * tests/check/elements/gdppay.c:
26873 remove declaration of buffers, it's already done in gstcheck.h
26875 2006-07-26 10:31:06 +0000 Tim-Philipp Müller <tim@centricular.net>
26877 gst/playback/: Remove GLib-2.6 compatibility cruft.
26878 Original commit message from CVS:
26879 * gst/playback/gstplaybasebin.c: (gst_play_base_bin_get_property):
26880 * gst/playback/gstplaybin.c: (gst_play_bin_get_property):
26881 Remove GLib-2.6 compatibility cruft.
26883 2006-07-24 16:47:10 +0000 Wim Taymans <wim.taymans@gmail.com>
26885 gst-libs/gst/audio/gstbaseaudiosink.c: Don't try to align a sample to an unknown value.
26886 Original commit message from CVS:
26887 * gst-libs/gst/audio/gstbaseaudiosink.c:
26888 (gst_base_audio_sink_render):
26889 Don't try to align a sample to an unknown value.
26891 2006-07-24 15:14:17 +0000 Wim Taymans <wim.taymans@gmail.com>
26893 gst-libs/gst/audio/gstbaseaudiosink.c: When the audio clock is slaved to another clock, never try to align samples bu...
26894 Original commit message from CVS:
26895 * gst-libs/gst/audio/gstbaseaudiosink.c:
26896 (gst_base_audio_sink_provide_clock), (gst_base_audio_sink_render):
26897 When the audio clock is slaved to another clock, never try to align
26898 samples but trust the rate interpolation algorithm.
26900 2006-07-24 14:34:42 +0000 Wim Taymans <wim.taymans@gmail.com>
26902 ext/alsa/gstalsasink.c: Don't try to calculate silence samples, base class does this much better now.
26903 Original commit message from CVS:
26904 * ext/alsa/gstalsasink.c: (set_hwparams), (gst_alsasink_prepare):
26905 Don't try to calculate silence samples, base class does this much
26907 * gst-libs/gst/audio/gstringbuffer.c: (build_linear_format),
26908 (gst_ring_buffer_debug_spec_caps), (gst_ring_buffer_parse_caps),
26909 (gst_ring_buffer_acquire):
26910 Calculate silence samples correctly.
26911 * gst-libs/gst/audio/gstringbuffer.h:
26914 2006-07-22 17:01:12 +0000 Tim-Philipp Müller <tim@centricular.net>
26916 gst/typefind/gsttypefindfunctions.c: Limit search for the first markup tag to the first few kB of the file. If we don...
26917 Original commit message from CVS:
26918 * gst/typefind/gsttypefindfunctions.c: (xml_check_first_element):
26919 Limit search for the first markup tag to the first few kB of
26920 the file. If we don't find one there, it's highly unlikely that
26921 this is an XML(-ish) file.
26923 2006-07-21 17:04:06 +0000 Andy Wingo <wingo@pobox.com>
26925 tests/check/pipelines/theoraenc.c (test_discontinuity): Similar test to the one in vorbisenc. Also commented out.
26926 Original commit message from CVS:
26927 2006-07-21 Andy Wingo <wingo@pobox.com>
26928 * tests/check/pipelines/theoraenc.c (test_discontinuity): Similar
26929 test to the one in vorbisenc. Also commented out.
26931 2006-07-21 16:54:19 +0000 Andy Wingo <wingo@pobox.com>
26933 tests/check/pipelines/vorbisenc.c: New test, commented out until Mike lands some elite vorbisenc patches.
26934 Original commit message from CVS:
26935 2006-07-21 Andy Wingo <wingo@pobox.com>
26936 * tests/check/pipelines/vorbisenc.c:
26937 (test_discontinuity): New test, commented out until Mike lands
26938 some elite vorbisenc patches.
26940 2006-07-21 15:59:24 +0000 Andy Wingo <wingo@pobox.com>
26942 tests/check/pipelines/: Port to bufferstraw.
26943 Original commit message from CVS:
26944 2006-07-21 Andy Wingo <wingo@pobox.com>
26945 * tests/check/pipelines/vorbisenc.c:
26946 * tests/check/pipelines/theoraenc.c: Port to bufferstraw.
26947 Bufferstraw was actually factored out of these tests. Now we share
26950 2006-07-21 11:03:28 +0000 Wim Taymans <wim.taymans@gmail.com>
26952 ext/theora/theoradec.c: Better clipping.
26953 Original commit message from CVS:
26954 * ext/theora/theoradec.c: (clip_buffer):
26957 2006-07-21 10:43:54 +0000 Wim Taymans <wim.taymans@gmail.com>
26959 gst-libs/gst/audio/gstaudiosink.c: Fix leak.
26960 Original commit message from CVS:
26961 * gst-libs/gst/audio/gstaudiosink.c: (audioringbuffer_thread_func),
26962 (gst_audioringbuffer_finalize), (gst_audioringbuffer_acquire),
26963 (gst_audioringbuffer_release), (gst_audioringbuffer_stop):
26965 Avoid type casting when we can.
26966 * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_dispose):
26969 2006-07-20 16:57:29 +0000 Tim-Philipp Müller <tim@centricular.net>
26971 ext/alsa/gstalsamixerelement.c: Make state change fail if the specified device can't be opened for some reason.
26972 Original commit message from CVS:
26973 * ext/alsa/gstalsamixerelement.c:
26974 (gst_alsa_mixer_element_change_state):
26975 Make state change fail if the specified device can't be opened
26978 2006-07-20 10:42:21 +0000 Wim Taymans <wim.taymans@gmail.com>
26980 gst/playback/test.c: Example of a small audio/video player using decodebin.
26981 Original commit message from CVS:
26982 * gst/playback/test.c: (gen_video_element), (gen_audio_element),
26983 (cb_newpad), (main):
26984 Example of a small audio/video player using decodebin.
26986 2006-07-20 05:56:48 +0000 Stefan Kost <ensonic@users.sourceforge.net>
26988 gst-libs/gst/riff/riff-ids.h: Add 'fact' chunk id
26989 Original commit message from CVS:
26990 * gst-libs/gst/riff/riff-ids.h:
26991 Add 'fact' chunk id
26993 2006-07-19 18:20:43 +0000 Wim Taymans <wim.taymans@gmail.com>
26995 gst-libs/gst/rtp/gstbasertpdepayload.c: Don't assert when not negotiated but post a meaningfull error message. Fixes ...
26996 Original commit message from CVS:
26997 * gst-libs/gst/rtp/gstbasertpdepayload.c:
26998 (gst_base_rtp_depayload_chain),
26999 (gst_base_rtp_depayload_change_state):
27000 Don't assert when not negotiated but post a meaningfull
27001 error message. Fixes #347918.
27002 * gst-libs/gst/rtp/gstbasertppayload.c:
27003 Add comment about better default MTU size.
27004 * gst-libs/gst/rtp/gstrtpbuffer.c: (gst_rtp_buffer_validate_data):
27005 Small cleanups, start docs.
27007 2006-07-19 14:46:36 +0000 Martin Szulecki <compiz@sukimashita.com>
27009 sys/v4l/gstv4lelement.c: If "device-name" is requested and the device is not open, try to temporarily open it to obta...
27010 Original commit message from CVS:
27011 Patch by: Martin Szulecki
27012 * sys/v4l/gstv4lelement.c: (gst_v4lelement_get_property):
27013 If "device-name" is requested and the device is not
27014 open, try to temporarily open it to obtain this
27015 information (#342494).
27017 2006-07-19 12:25:00 +0000 Tim-Philipp Müller <tim@centricular.net>
27019 gst-libs/gst/tag/gstid3tag.c: Add TSSE <=> GST_TAG_ENCODER mapping (see #347898).
27020 Original commit message from CVS:
27021 * gst-libs/gst/tag/gstid3tag.c:
27022 Add TSSE <=> GST_TAG_ENCODER mapping (see #347898).
27023 * gst-libs/gst/tag/gsttageditingprivate.h:
27024 * gst-libs/gst/tag/gstvorbistag.c:
27025 Some more random const-ifications.
27027 2006-07-18 19:48:48 +0000 Stefan Kost <ensonic@users.sourceforge.net>
27029 gst-libs/gst/riff/: add comment what those 16 bytes in struct _gst_riff_strh according to one avi-dumper are
27030 Original commit message from CVS:
27031 * gst-libs/gst/riff/riff-ids.h:
27032 * gst-libs/gst/riff/riff-media.c:
27033 (gst_riff_create_video_template_caps):
27034 Add more FOURCCs (sort list to make stuff easier to find),
27035 add comment what those 16 bytes in struct _gst_riff_strh according to
27038 2006-07-17 14:17:45 +0000 Thomas Vander Stichele <thomas@apestaart.org>
27040 gst/gdp/: remove parent_class setting, BOILERPLATE does this fix typo in comment
27041 Original commit message from CVS:
27042 2006-07-17 Thomas Vander Stichele <thomas at apestaart dot org>
27043 * gst/gdp/gstgdpdepay.c: (gst_gdp_depay_class_init):
27044 * gst/gdp/gstgdppay.c: (gst_gdp_pay_class_init),
27045 remove parent_class setting, BOILERPLATE does this
27046 (gst_gdp_pay_reset_streamheader):
27047 fix typo in comment
27049 2006-07-17 13:48:10 +0000 Tim-Philipp Müller <tim@centricular.net>
27051 gst-libs/gst/audio/multichannel.c: Const-ify two arrays.
27052 Original commit message from CVS:
27053 * gst-libs/gst/audio/multichannel.c:
27054 (gst_audio_check_channel_positions),
27055 (gst_audio_fixate_channel_positions):
27056 Const-ify two arrays.
27058 2006-07-17 12:33:42 +0000 Tim-Philipp Müller <tim@centricular.net>
27060 ext/alsa/gstalsa.c: Fix typo, so that alsasink also advertises 8 channels if that's supported (tags: can, worms, open...
27061 Original commit message from CVS:
27062 * ext/alsa/gstalsa.c: (caps_add_channel_configuration):
27063 Fix typo, so that alsasink also advertises 8 channels
27064 if that's supported (tags: can, worms, open, alsa, ph34r).
27066 2006-07-17 12:01:04 +0000 Wim Taymans <wim.taymans@gmail.com>
27068 ext/ogg/gstoggdemux.c: *sigh*, when is the compiler going to warn when the comments are out-of-sync with the code.. R...
27069 Original commit message from CVS:
27070 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_internal_chain),
27071 (gst_ogg_pad_submit_packet), (gst_ogg_demux_read_chain):
27072 *sigh*, when is the compiler going to warn when the comments
27073 are out-of-sync with the code.. Refix case of busted theora
27074 headers with 0 granule pos.
27076 2006-07-14 17:56:59 +0000 Wim Taymans <wim.taymans@gmail.com>
27078 gst-libs/gst/rtp/gstbasertpdepayload.c: Fix 99% cpu load by waiting for absolute times on the clock. Fixes #347300.
27079 Original commit message from CVS:
27080 * gst-libs/gst/rtp/gstbasertpdepayload.c:
27081 (gst_base_rtp_depayload_wait),
27082 (gst_base_rtp_depayload_change_state),
27083 (gst_base_rtp_depayload_set_property),
27084 (gst_base_rtp_depayload_get_property):
27085 Fix 99% cpu load by waiting for absolute times on the
27086 clock. Fixes #347300.
27088 2006-07-14 17:07:08 +0000 Andy Wingo <wingo@pobox.com>
27090 ext/theora/gsttheoraparse.h: ext/theora/theoraparse.c (theora_parse_drain_event_queue) (theora_parse_push_headers, th...
27091 Original commit message from CVS:
27092 2006-07-14 Andy Wingo <wingo@pobox.com>
27093 * ext/theora/gsttheoraparse.h:
27094 * ext/theora/theoraparse.c (theora_parse_drain_event_queue)
27095 (theora_parse_push_headers, theora_parse_clear_queue)
27096 (theora_parse_drain_queue_prematurely, )
27097 (theora_parse_sink_event, theora_parse_change_state): Queue events
27098 until we initialized our state, like in vorbisparse.
27100 2006-07-14 16:45:17 +0000 Iain * <iaingnome@gmail.com>
27102 ext/vorbis/vorbisparse.h: ext/vorbis/vorbisparse.c (vorbis_parse_drain_event_queue) (vorbis_parse_push_headers, vorbi...
27103 Original commit message from CVS:
27104 2006-07-14 Andy Wingo <wingo@pobox.com>
27105 * ext/vorbis/vorbisparse.h:
27106 * ext/vorbis/vorbisparse.c (vorbis_parse_drain_event_queue)
27107 (vorbis_parse_push_headers, vorbis_parse_clear_queue)
27108 (vorbis_parse_drain_queue_prematurely, )
27109 (vorbis_parse_sink_event, vorbis_parse_change_state): Queue events
27110 until we have initialized our state. Fixes seeking after an
27112 2006-07-14 Andy Wingo <wingo@pobox.com>
27113 Patch by: Iain * <iaingnome@gmail.com>
27114 * ext/ogg/gstoggdemux.c (gst_ogg_demux_finalize): Fix memleak.
27116 2006-07-14 15:52:39 +0000 Jan Schmidt <thaytan@mad.scientist.com>
27118 configure.ac: Bump nano back to CVS
27119 Original commit message from CVS:
27121 Bump nano back to CVS
27123 === release 0.10.9 ===
27125 2006-07-14 15:51:24 +0000 Jan Schmidt <thaytan@mad.scientist.com>
27127 configure.ac: releasing 0.10.9, "I walk the line"
27128 Original commit message from CVS:
27129 2006-07-13 Jan Schmidt <thaytan@mad.scientist.com>
27131 releasing 0.10.9, "I walk the line"
27133 2006-07-14 14:12:40 +0000 Michael Smith <msmith@xiph.org>
27135 tests/check/pipelines/vorbisenc.c: Move a g_cond_signal to earlier to avoid sometimes deadlocking (commonly happens w...
27136 Original commit message from CVS:
27137 * tests/check/pipelines/vorbisenc.c: (stop_pipeline):
27138 Move a g_cond_signal to earlier to avoid sometimes deadlocking
27139 (commonly happens when running this test under valgrind) when trying
27140 to remove the buffer probe.
27142 2006-07-14 10:34:36 +0000 Thomas Vander Stichele <thomas@apestaart.org>
27144 * gst/gdp/Makefile.am:
27145 build as a plugin, not a lib
27146 Original commit message from CVS:
27147 build as a plugin, not a lib
27149 2006-07-13 16:43:22 +0000 Jan Schmidt <thaytan@mad.scientist.com>
27151 sys/xvimage/xvimagesink.c: Fix missing g_unlock from the previous commit
27152 Original commit message from CVS:
27153 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_new):
27154 Fix missing g_unlock from the previous commit
27156 2006-07-13 16:34:04 +0000 Jan Schmidt <thaytan@mad.scientist.com>
27158 sys/: Implement a locking order to ensure we always take the object lock before the x_lock and never vice-versa.
27159 Original commit message from CVS:
27160 * sys/ximage/ximagesink.c: (gst_ximagesink_ximage_new),
27161 (gst_ximagesink_change_state):
27162 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_new),
27163 (gst_xvimagesink_change_state):
27164 Implement a locking order to ensure we always take the object lock
27165 before the x_lock and never vice-versa.
27167 2006-07-13 15:25:58 +0000 Thomas Vander Stichele <thomas@apestaart.org>
27169 docs/plugins/: add more plugins and elements to docs
27170 Original commit message from CVS:
27171 * docs/plugins/Makefile.am:
27172 * docs/plugins/gst-plugins-bad-plugins-docs.sgml:
27173 * docs/plugins/gst-plugins-bad-plugins-sections.txt:
27174 add more plugins and elements to docs
27175 * gst/gdp/gstgdpdepay.c: (gst_gdp_depay_chain):
27176 fix segfaults due to wrong g_free
27178 * gst/gdp/gstgdppay.c:
27181 2006-07-13 14:38:15 +0000 Jan Schmidt <thaytan@mad.scientist.com>
27183 gst/playback/gstdecodebin.c: Fix a caps leak when linking (#347304)
27184 Original commit message from CVS:
27185 * gst/playback/gstdecodebin.c: (find_compatibles):
27186 Fix a caps leak when linking (#347304)
27187 * sys/ximage/ximagesink.c: (gst_ximage_buffer_finalize),
27188 (gst_ximagesink_ximage_destroy), (gst_ximagesink_xcontext_clear),
27189 (gst_ximagesink_change_state):
27190 * sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_destroy),
27191 (gst_xvimage_buffer_finalize), (gst_xvimagesink_check_xshm_calls),
27192 (gst_xvimagesink_xvimage_new), (gst_xvimagesink_xvimage_put),
27193 (gst_xvimagesink_xcontext_clear), (gst_xvimagesink_change_state):
27194 Don't leak shared memory resources. Use the object lock to protect
27195 against the xcontext disappearing while returning a buffer from the
27196 pipeline. (#347304)
27198 2006-07-12 14:20:43 +0000 Edward Hervey <bilboed@bilboed.com>
27200 ext/vorbis/vorbisdec.c: gst_tag_list_merge() returns a new object. Take that into account when using it. This avoids ...
27201 Original commit message from CVS:
27202 * ext/vorbis/vorbisdec.c: (vorbis_dec_finalize),
27203 (vorbis_handle_comment_packet):
27204 gst_tag_list_merge() returns a new object. Take that into account when
27205 using it. This avoids memleak.
27206 Revert previous commit which is not needed.
27208 2006-07-12 13:30:20 +0000 Edward Hervey <bilboed@bilboed.com>
27210 ext/vorbis/vorbisdec.c: Reset the decoder in finalize so that all fields get cleared.
27211 Original commit message from CVS:
27212 * ext/vorbis/vorbisdec.c: (vorbis_dec_finalize):
27213 Reset the decoder in finalize so that all fields get cleared.
27215 2006-07-12 13:24:19 +0000 Wim Taymans <wim.taymans@gmail.com>
27217 gst-libs/gst/audio/gstbaseaudiosrc.c: Don't try to post an error message when setting the clock fails as this can hap...
27218 Original commit message from CVS:
27219 * gst-libs/gst/audio/gstbaseaudiosrc.c:
27220 (gst_base_audio_src_set_clock),
27221 (gst_base_audio_src_check_get_range), (gst_base_audio_src_create):
27222 Don't try to post an error message when setting the clock fails
27223 as this can happen when adding an element to a bin which will then
27224 deadlock. Fixes #347296.
27226 2006-07-12 13:04:15 +0000 Edward Hervey <bilboed@bilboed.com>
27228 ext/vorbis/vorbisdec.c: Post tag messages on the bus even if we're not initialized.
27229 Original commit message from CVS:
27230 * ext/vorbis/vorbisdec.c: (gst_vorbis_dec_reset),
27231 (vorbis_dec_sink_event), (vorbis_handle_comment_packet),
27232 (vorbis_handle_type_packet):
27233 Post tag messages on the bus even if we're not initialized.
27234 If we're not initialized, we still postpone the event pushing of tags.
27236 2006-07-12 11:28:37 +0000 Wim Taymans <wim.taymans@gmail.com>
27238 Revert last two changes that broke the freeze.
27239 Original commit message from CVS:
27240 * ext/alsa/gstalsasink.c: (set_hwparams), (gst_alsasink_prepare):
27241 * gst-libs/gst/audio/gstringbuffer.c: (build_linear_format),
27242 (gst_ring_buffer_debug_spec_caps), (gst_ring_buffer_parse_caps):
27243 Revert last two changes that broke the freeze.
27245 2006-07-12 10:59:55 +0000 Wim Taymans <wim.taymans@gmail.com>
27247 ext/alsa/gstalsasink.c: basesink calculates silence sample correctly for us.
27248 Original commit message from CVS:
27249 * ext/alsa/gstalsasink.c: (set_hwparams), (gst_alsasink_prepare):
27250 basesink calculates silence sample correctly for us.
27252 2006-07-12 10:58:42 +0000 Wim Taymans <wim.taymans@gmail.com>
27254 gst-libs/gst/audio/gstringbuffer.c: Calculate correct silence samples so we don't fill our ringbuffer with noise.
27255 Original commit message from CVS:
27256 * gst-libs/gst/audio/gstringbuffer.c: (build_linear_format),
27257 (gst_ring_buffer_debug_spec_caps), (gst_ring_buffer_parse_caps):
27258 Calculate correct silence samples so we don't fill our ringbuffer
27261 2006-07-12 10:37:18 +0000 Edward Hervey <bilboed@bilboed.com>
27263 ext/vorbis/vorbisdec.*: Delay sending events (newsegment, tags) until the decoder is properly initialized.
27264 Original commit message from CVS:
27265 * ext/vorbis/vorbisdec.c: (gst_vorbis_dec_init),
27266 (gst_vorbis_dec_reset), (vorbis_dec_sink_event),
27267 (vorbis_handle_comment_packet), (vorbis_handle_type_packet):
27268 * ext/vorbis/vorbisdec.h:
27269 Delay sending events (newsegment, tags) until the decoder is properly
27273 2006-07-11 22:40:13 +0000 Jan Schmidt <thaytan@mad.scientist.com>
27290 Original commit message from CVS:
27293 2006-07-11 21:04:16 +0000 Jan Schmidt <thaytan@mad.scientist.com>
27295 tests/check/elements/audioconvert.c: Patch from #347221 adding a test for audioconvert channel remappings.
27296 Original commit message from CVS:
27297 * tests/check/elements/audioconvert.c: (get_float_mc_caps),
27298 (get_int_mc_caps), (GST_START_TEST), (audioconvert_suite):
27299 Patch from #347221 adding a test for audioconvert
27300 channel remappings.
27302 2006-07-11 12:03:25 +0000 Tim-Philipp Müller <tim@centricular.net>
27304 gst/subparse/gstssaparse.c: Don't include the terminating NUL in the buffer size, it's only there for extra paranoia ...
27305 Original commit message from CVS:
27306 * gst/subparse/gstssaparse.c: (gst_ssa_parse_base_init),
27307 (gst_ssa_parse_parse_line):
27308 Don't include the terminating NUL in the buffer size,
27309 it's only there for extra paranoia (would add random
27310 '*' characters at the end of each subtitle since the
27311 terminator itself is not valid UTF-8 technically).
27312 Also fix indenting after boilerplate macro.
27314 2006-07-10 14:59:03 +0000 Tim-Philipp Müller <tim@centricular.net>
27316 gst/playback/gstdecodebin.c: Also emit 'unknown-type' signal (which should really be called unhandled-type) if we fou...
27317 Original commit message from CVS:
27318 * gst/playback/gstdecodebin.c: (close_pad_link):
27319 Also emit 'unknown-type' signal (which should really be
27320 called unhandled-type) if we found potential decoders/demuxers
27321 in the registry but none of them worked in the end (as in the
27322 case where the plugins don't exist any longer but are still
27323 listed in the registry). Fixes #329798.
27325 2006-07-08 13:48:58 +0000 Andy Wingo <wingo@pobox.com>
27328 * ext/theora/theoraparse.c:
27329 theoraparse.c (theora_parse_push_buffer)
27330 Original commit message from CVS:
27331 2006-07-08 Andy Wingo <wingo@pobox.com>
27332 * theoraparse.c (theora_parse_push_buffer)
27333 (theora_parse_drain_queue_prematurely, theora_parse_drain_queue):
27334 Add some more debugging. Fix granulepos reconstruction in the face
27335 of discontinuities.
27337 2006-07-06 15:54:50 +0000 Wim Taymans <wim.taymans@gmail.com>
27339 gst-libs/gst/audio/gstbaseaudiosink.c: Use gobject_class instead of G_OBJECT_CLASS (klass)
27340 Original commit message from CVS:
27341 * gst-libs/gst/audio/gstbaseaudiosink.c:
27342 (gst_base_audio_sink_class_init),
27343 (gst_base_audio_sink_provide_clock):
27344 Use gobject_class instead of G_OBJECT_CLASS (klass)
27345 * gst-libs/gst/audio/gstbaseaudiosrc.c:
27346 (gst_base_audio_src_class_init), (gst_base_audio_src_init),
27347 (gst_base_audio_src_set_clock), (gst_base_audio_src_provide_clock),
27348 (gst_base_audio_src_get_time),
27349 (gst_base_audio_src_check_get_range), (gst_base_audio_src_create),
27350 (gst_base_audio_src_create_ringbuffer):
27351 Fix latency and buffer-time constants and properties ala basesink.
27352 Implement pull based scheduling. Fixes #346527.
27353 Set default blocksize in GstBaseSrc to 0, we default to pushing out
27355 Refuse slaving to another clock instead of silently not working.
27356 Only provide a clock when we are actually able to do so.
27357 Various small cleanups and compiler hints.
27359 2006-07-06 13:23:07 +0000 Lutz Mueller <lutz@topfrose.de>
27361 gst/typefind/gsttypefindfunctions.c: Add typefinding for text/html (#346581).
27362 Original commit message from CVS:
27363 Patch by: Lutz Mueller <lutz at topfrose de>
27364 * gst/typefind/gsttypefindfunctions.c: (html_type_find),
27366 Add typefinding for text/html (#346581).
27368 2006-07-06 13:12:02 +0000 Tim-Philipp Müller <tim@centricular.net>
27370 gst/typefind/gsttypefindfunctions.c: Fix SMIL typefinding, make xml_check_first_element() more useful.
27371 Original commit message from CVS:
27372 * gst/typefind/gsttypefindfunctions.c: (utf8_type_find),
27373 (xml_check_first_element), (xml_type_find), (smil_type_find):
27374 Fix SMIL typefinding, make xml_check_first_element() more
27377 2006-07-06 13:04:24 +0000 Tim-Philipp Müller <tim@centricular.net>
27379 gst/playback/gstplaybasebin.*: Protect list of elements with a subtitle-encoding property and the subtitle encoding m...
27380 Original commit message from CVS:
27381 * gst/playback/gstplaybasebin.c: (gst_play_base_bin_init),
27382 (gst_play_base_bin_finalize), (decodebin_element_added_cb),
27383 (decodebin_element_removed_cb), (gst_play_base_bin_set_property):
27384 * gst/playback/gstplaybasebin.h:
27385 Protect list of elements with a subtitle-encoding property and
27386 the subtitle encoding member itself with a lock of their own
27387 instead of using the object lock. This prevents a dead-lock in
27388 the element-remove callback in some circumstances when shutting
27391 2006-07-05 20:11:13 +0000 Sébastien Moutte <sebastien@moutte.net>
27393 win32/common/libgsttag.def: Export some new functions.
27394 Original commit message from CVS:
27395 * win32/common/libgsttag.def:
27396 Export some new functions.
27397 * win32/vs6/libgstogg.dsp:
27398 Add a link to libgsttag-0.10.lib.
27400 2006-07-04 16:50:21 +0000 Tim-Philipp Müller <tim@centricular.net>
27402 ext/alsa/gstalsamixertrack.c: Some const-ification.
27403 Original commit message from CVS:
27404 * ext/alsa/gstalsamixertrack.c: (gst_alsa_mixer_track_new):
27405 Some const-ification.
27407 2006-07-04 14:06:03 +0000 Wim Taymans <wim.taymans@gmail.com>
27409 gst/playback/gstplaybasebin.c: Improve checking if we are dealing with a stream. Added some more uris that need buffe...
27410 Original commit message from CVS:
27411 * gst/playback/gstplaybasebin.c: (is_stream), (gen_source_element):
27412 Improve checking if we are dealing with a stream. Added some
27413 more uris that need buffering.
27415 2006-07-03 10:43:31 +0000 Edward Hervey <bilboed@bilboed.com>
27417 ext/vorbis/vorbisdec.c: Remove unused variable.
27418 Original commit message from CVS:
27419 * ext/vorbis/vorbisdec.c: (vorbis_do_clip):
27420 Remove unused variable.
27422 2006-07-02 21:48:51 +0000 Thomas Vander Stichele <thomas@apestaart.org>
27424 Makefile.am: include lcov.mak
27425 Original commit message from CVS:
27429 add GCOV_LIBS to GST_LIBS
27431 2006-07-02 11:08:58 +0000 Michael Sheldon <webmaster@mikeasoft.com>
27433 ext/alsa/gstalsasrc.c: Add 32 bps to template caps and increase channels range from [1,2] to [1,MAX]. See #346326.
27434 Original commit message from CVS:
27435 Patch by: Michael Sheldon <webmaster at mikeasoft com>
27436 * ext/alsa/gstalsasrc.c:
27437 Add 32 bps to template caps and increase channels range
27438 from [1,2] to [1,MAX]. See #346326.
27440 2006-06-30 12:04:51 +0000 Tim-Philipp Müller <tim@centricular.net>
27442 gst-libs/gst/riff/riff-media.c: Recognise 'WMVA' video codec fourcc (#345879).
27443 Original commit message from CVS:
27444 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
27445 Recognise 'WMVA' video codec fourcc (#345879).
27447 2006-06-29 12:21:06 +0000 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
27449 gst-libs/gst/rtp/gstbasertpaudiopayload.c: Fixed nasty memory leak
27450 Original commit message from CVS:
27451 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
27452 Fixed nasty memory leak
27454 2006-06-26 13:19:33 +0000 Thomas Vander Stichele <thomas@apestaart.org>
27456 gst/tcp/gsttcp.c: fix logging
27457 Original commit message from CVS:
27458 * gst/tcp/gsttcp.c: (gst_tcp_read_buffer),
27459 (gst_tcp_gdp_read_buffer), (gst_tcp_gdp_read_caps):
27462 2006-06-23 16:45:50 +0000 Jan Schmidt <thaytan@mad.scientist.com>
27464 gst/playback/gstdecodebin.c: Protect remove_fakesink using a mutex, so that we don't try and remove the fakesink simu...
27465 Original commit message from CVS:
27466 * gst/playback/gstdecodebin.c: (gst_decode_bin_class_init),
27467 (gst_decode_bin_init), (gst_decode_bin_finalize), (add_fakesink),
27468 (remove_fakesink), (pad_probe), (gst_decode_bin_change_state):
27469 Protect remove_fakesink using a mutex, so that we don't try and
27470 remove the fakesink simultaneously from multiple threads.
27471 When going from READY to PAUSED, restore the fakesink, so that
27472 it is there when decodebin gets reused.
27474 2006-06-23 09:53:09 +0000 Tim-Philipp Müller <tim@centricular.net>
27476 Use GST_DEBUG_CATEGORY_STATIC where possible (#342503).
27477 Original commit message from CVS:
27478 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
27479 * gst-libs/gst/rtp/gstbasertpdepayload.c:
27480 * gst-libs/gst/rtp/gstbasertppayload.c:
27481 * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
27482 * gst/tcp/gstmultifdsink.c:
27483 * gst/tcp/gsttcpclientsink.c:
27484 * gst/tcp/gsttcpclientsrc.c:
27485 * gst/tcp/gsttcpserversink.c:
27486 * gst/tcp/gsttcpserversrc.c:
27487 * gst/videorate/gstvideorate.c:
27488 * gst/videotestsrc/gstvideotestsrc.c:
27489 * sys/v4l/gstv4ljpegsrc.c:
27490 * sys/v4l/gstv4lmjpegsink.c:
27491 * sys/v4l/gstv4lsrc.c:
27492 * tests/examples/seek/scrubby.c:
27493 * tests/examples/seek/seek.c:
27494 Use GST_DEBUG_CATEGORY_STATIC where possible (#342503).
27496 2006-06-23 09:09:44 +0000 Tim-Philipp Müller <tim@centricular.net>
27498 Use GST_DEBUG_CATEGORY_STATIC where possible (#342503) and fix one GObject boilerplate macro.
27499 Original commit message from CVS:
27500 * ext/directfb/dfbvideosink.c:
27501 * ext/gsm/gstgsmdec.c:
27502 * ext/gsm/gstgsmenc.c:
27503 * ext/libmms/gstmms.c:
27504 * ext/neon/gstneonhttpsrc.c:
27505 * ext/theora/theoradec.c:
27506 * gst/freeze/gstfreeze.c:
27507 * gst/gdp/gstgdpdepay.c:
27508 * gst/gdp/gstgdppay.c:
27509 * sys/glsink/glimagesink.c:
27510 Use GST_DEBUG_CATEGORY_STATIC where possible (#342503)
27511 and fix one GObject boilerplate macro.
27513 2006-06-22 12:13:31 +0000 Tim-Philipp Müller <tim@centricular.net>
27515 gst-libs/gst/tag/tags.c: Second field in GEnumValue shouldn't be a description, but a stringified version of the enum...
27516 Original commit message from CVS:
27517 * gst-libs/gst/tag/tags.c: (register_tag_image_type_enum):
27518 Second field in GEnumValue shouldn't be a description,
27519 but a stringified version of the enum value.
27521 2006-06-22 12:03:14 +0000 Wim Taymans <wim.taymans@gmail.com>
27523 sys/ximage/ximagesink.c: Avoid type checking in buffer casts.
27524 Original commit message from CVS:
27525 * sys/ximage/ximagesink.c: (gst_ximage_buffer_finalize),
27526 (gst_ximage_buffer_free), (gst_ximagesink_ximage_put),
27527 (gst_ximagesink_setcaps), (gst_ximagesink_buffer_alloc):
27528 Avoid type checking in buffer casts.
27529 Avoid caps copy in buffer_alloc when we can.
27530 Use pad_peer_accept.
27532 2006-06-22 11:01:41 +0000 Tim-Philipp Müller <tim@centricular.net>
27534 gst-libs/gst/tag/tag.h: Oops, make that 'Since: 0.10.9'.
27535 Original commit message from CVS:
27536 * gst-libs/gst/tag/tag.h:
27537 Oops, make that 'Since: 0.10.9'.
27539 2006-06-22 10:55:05 +0000 Tim-Philipp Müller <tim@centricular.net>
27541 API: add GstTagImageType enum to describe images contained in image tags (#345641).
27542 Original commit message from CVS:
27543 * docs/libs/gst-plugins-base-libs-sections.txt:
27544 * gst-libs/gst/tag/tag.h:
27545 * gst-libs/gst/tag/tags.c: (register_tag_image_type_enum),
27546 (gst_tag_image_type_get_type):
27547 API: add GstTagImageType enum to describe images contained
27548 in image tags (#345641).
27550 2006-06-22 10:31:22 +0000 Tim-Philipp Müller <tim@centricular.net>
27552 gst/tcp/gstmultifdsink.c: Fix warnings with gst-inspect: "buffers-min" property should be of G_TYPE_INT and not G_TYP...
27553 Original commit message from CVS:
27554 * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init):
27555 Fix warnings with gst-inspect: "buffers-min" property
27556 should be of G_TYPE_INT and not G_TYPE_INT64. Also fix
27557 typo in property description.
27559 2006-06-22 10:10:51 +0000 Cody Russell <bratsche@gnome.org>
27561 gst/: Avoid unnecessary class cast check in class_init functions (#337747).
27562 Original commit message from CVS:
27563 Patch by: Cody Russell <bratsche at gnome org>
27564 * gst/audioresample/gstaudioresample.c:
27565 (gst_audioresample_class_init):
27566 * gst/playback/gststreamselector.c:
27567 (gst_stream_selector_class_init):
27568 * gst/subparse/gstsubparse.c: (gst_sub_parse_class_init):
27569 * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init):
27570 * gst/tcp/gsttcpclientsink.c: (gst_tcp_client_sink_class_init):
27571 * gst/tcp/gsttcpclientsrc.c: (gst_tcp_client_src_class_init):
27572 * gst/tcp/gsttcpserversink.c: (gst_tcp_server_sink_class_init):
27573 * gst/tcp/gsttcpserversrc.c: (gst_tcp_server_src_class_init):
27574 * gst/videoscale/gstvideoscale.c: (gst_video_scale_class_init):
27575 * gst/videotestsrc/gstvideotestsrc.c:
27576 (gst_video_test_src_class_init):
27577 * gst/volume/gstvolume.c: (gst_volume_class_init):
27578 Avoid unnecessary class cast check in class_init
27579 functions (#337747).
27581 2006-06-21 18:39:07 +0000 Tim-Philipp Müller <tim@centricular.net>
27583 ext/pango/gsttextoverlay.c: g_markup_escape_text() REALLY doesn't like non-UTF8 input and doesn't validate its input ...
27584 Original commit message from CVS:
27585 * ext/pango/gsttextoverlay.c: (gst_text_overlay_make_utf8),
27586 (gst_text_overlay_video_chain):
27587 g_markup_escape_text() REALLY doesn't like non-UTF8 input
27588 and doesn't validate its input either (and neither did
27589 textoverlay it seems). Let's do that then and fix #345206.
27591 2006-06-19 17:12:57 +0000 Wim Taymans <wim.taymans@gmail.com>
27593 gst/tcp/gstmultifdsink.*: Added shiny new burst-on-connect methods.
27594 Original commit message from CVS:
27595 * gst/tcp/gstmultifdsink.c: (gst_sync_method_get_type),
27596 (gst_unit_type_get_type), (gst_multi_fd_sink_class_init),
27597 (gst_multi_fd_sink_init), (gst_multi_fd_sink_add_full),
27598 (gst_multi_fd_sink_add), (gst_multi_fd_sink_handle_client_read),
27599 (find_syncframe), (find_limits), (assign_value),
27600 (count_burst_unit), (gst_multi_fd_sink_new_client),
27601 (gst_multi_fd_sink_handle_client_write),
27602 (gst_multi_fd_sink_queue_buffer), (gst_multi_fd_sink_render),
27603 (gst_multi_fd_sink_set_property), (gst_multi_fd_sink_get_property),
27604 (gst_multi_fd_sink_change_state):
27605 * gst/tcp/gstmultifdsink.h:
27606 Added shiny new burst-on-connect methods.
27607 Add properties to control the minimal amount of data queued.
27609 API: bytes-min property
27610 API: time-min property
27611 API: buffers-min property
27612 API: burst-unit property
27613 API: burst-value property
27614 API: add-full signal
27615 * gst/tcp/gsttcp-marshal.list:
27616 Added new marshaller code for the new signal.
27617 * tests/check/elements/multifdsink.c: (GST_START_TEST),
27618 (multifdsink_suite):
27619 Added testcases for new burst methods.
27621 2006-06-19 11:35:47 +0000 Christian Schaller <uraeus@gnome.org>
27623 * gst-plugins-base.spec.in:
27624 update for latest changes
27625 Original commit message from CVS:
27626 update for latest changes
27628 2006-06-19 09:57:50 +0000 Edward Hervey <bilboed@bilboed.com>
27630 ext/theora/theoradec.c: Implement clipping for accurate seeking.
27631 Original commit message from CVS:
27632 * ext/theora/theoradec.c: (clip_buffer), (theora_dec_push):
27633 Implement clipping for accurate seeking.
27636 2006-06-19 09:08:05 +0000 Philip Jaegenstedt <philip@lysator.liu.se>
27638 gst/videoscale/gstvideoscale.c: Make videoscale support RGBA, ARGB, BGRA and ABGR. Fixes #345131
27639 Original commit message from CVS:
27640 Patch by: Philip Jaegenstedt <philip at lysator dot liu dot se>
27641 * gst/videoscale/gstvideoscale.c: (gst_video_scale_prepare_size),
27642 (gst_video_scale_transform):
27643 Make videoscale support RGBA, ARGB, BGRA and ABGR. Fixes #345131
27645 2006-06-17 14:18:41 +0000 Tim-Philipp Müller <tim@centricular.net>
27649 Original commit message from CVS:
27652 2006-06-17 14:13:03 +0000 Tim-Philipp Müller <tim@centricular.net>
27654 configure.ac: Fix --disable-extern (can't set conditionals conditionally, #343602).
27655 Original commit message from CVS:
27657 Fix --disable-extern (can't set conditionals conditionally,
27660 2006-06-16 15:43:23 +0000 Tim-Philipp Müller <tim@centricular.net>
27662 tests/check/elements/audioresample.c: Add test case for bug #342789 fixed below.
27663 Original commit message from CVS:
27664 * tests/check/elements/audioresample.c: (test_reuse),
27665 (audioresample_suite):
27666 Add test case for bug #342789 fixed below.
27668 2006-06-16 15:17:44 +0000 Tim-Philipp Müller <tim@centricular.net>
27670 gst/audioresample/gstaudioresample.c: Implement GstBaseTransform::start and ::stop so that audioresample can clear it...
27671 Original commit message from CVS:
27672 * gst/audioresample/gstaudioresample.c:
27673 (gst_audioresample_class_init), (gst_audioresample_init),
27674 (audioresample_start), (audioresample_stop),
27675 (gst_audioresample_set_property), (gst_audioresample_get_property):
27676 Implement GstBaseTransform::start and ::stop so that audioresample
27677 can clear its internal state properly and be reused insted of
27678 causing non-negotiated errors with playbin under some circumstances
27680 * tests/check/elements/audioresample.c: (setup_audioresample),
27681 (cleanup_audioresample):
27682 Need to set element state here so that ::start and ::stop are
27685 2006-06-16 13:59:29 +0000 Young-Ho Cha <ganadist@chollian.net>
27687 gst-libs/gst/riff/riff-read.c: Parse extra data better, apparently it's right behind the normal strf header size. Fix...
27688 Original commit message from CVS:
27689 Patch by: Young-Ho Cha <ganadist at chollian dot net>
27690 * gst-libs/gst/riff/riff-read.c: (gst_riff_parse_strf_vids):
27691 Parse extra data better, apparently it's right behind
27692 the normal strf header size. Fixes #343500.
27694 2006-06-16 11:04:21 +0000 Wim Taymans <wim.taymans@gmail.com>
27696 ext/alsa/gstalsasink.c: If we fail to set the buffer_time and period_time alsa parameters, post a warning and leave a...
27697 Original commit message from CVS:
27698 * ext/alsa/gstalsasink.c: (set_hwparams):
27699 If we fail to set the buffer_time and period_time alsa
27700 parameters, post a warning and leave alsa select a
27701 default instead of failing. Fixes #342085
27703 2006-06-16 10:30:25 +0000 Tim-Philipp Müller <tim@centricular.net>
27706 ChangeLog surgery (it wouldn't have crashed, just shown bogus values)
27707 Original commit message from CVS:
27708 ChangeLog surgery (it wouldn't have crashed, just shown bogus values)
27710 2006-06-16 10:20:10 +0000 Tim-Philipp Müller <tim@centricular.net>
27712 Remove GST_CDDA_TAG_TRACK_TAGS again, it is #ifdef 0'ed out in the header file and shouldn't be listed in the docs.
27713 Original commit message from CVS:
27714 * docs/libs/gst-plugins-base-libs-sections.txt:
27715 * gst-libs/gst/cdda/gstcddabasesrc.h:
27716 Remove GST_CDDA_TAG_TRACK_TAGS again, it is #ifdef 0'ed
27717 out in the header file and shouldn't be listed in the docs.
27718 * gst-libs/gst/riff/riff-read.c: (gst_riff_read_chunk):
27719 Fix it so that it doesn't crash in the debug statement.
27721 2006-06-16 10:02:25 +0000 Stefan Kost <ensonic@users.sourceforge.net>
27723 docs/libs/: add remaining symbols into correct setions
27724 Original commit message from CVS:
27725 * docs/libs/Makefile.am:
27726 * docs/libs/gst-plugins-base-libs-docs.sgml:
27727 * docs/libs/gst-plugins-base-libs-sections.txt:
27728 * docs/libs/gst-plugins-base-libs.types:
27729 add remaining symbols into correct setions
27730 * gst-libs/gst/audio/gstringbuffer.c:
27731 fix incomplete docs
27732 * gst-libs/gst/audio/gstringbuffer.h:
27733 comment out not yet implemented function
27734 * gst-libs/gst/floatcast/floatcast.h:
27735 * gst-libs/gst/netbuffer/gstnetbuffer.c:
27736 add short descriptions
27737 * gst-libs/gst/interfaces/propertyprobe.c:
27738 fix return value docs
27739 * gst-libs/gst/riff/riff-read.c: (gst_riff_read_chunk):
27740 simplify debug logging
27741 * gst-libs/gst/riff/riff-read.h:
27742 sync function prototype and docs
27743 * gst-libs/gst/rtp/gstbasertpaudiopayload.h:
27744 remove left over symbol
27746 2006-06-16 09:56:41 +0000 Tim-Philipp Müller <tim@centricular.net>
27748 Use GST_PLUGIN_DOCS macro in configure.ac, add
27749 Original commit message from CVS:
27752 * docs/Makefile.am:
27753 Use GST_PLUGIN_DOCS macro in configure.ac, add
27754 --enable-plugin-docs default to autogen.sh and use
27755 ENABLE_PLUGIN_DOCS conditional in Makefile.am (#344039).
27757 2006-06-15 15:27:49 +0000 Wim Taymans <wim.taymans@gmail.com>
27759 ext/ogg/gstoggdemux.c: Combine GstFlowReturn from the source pads to give a meaningfull result to the upstream peer o...
27760 Original commit message from CVS:
27761 * ext/ogg/gstoggdemux.c: (gst_ogg_demux_chain_peer),
27762 (gst_ogg_demux_activate_chain), (gst_ogg_demux_combine_flows),
27763 (gst_ogg_demux_loop):
27764 Combine GstFlowReturn from the source pads to give a
27765 meaningfull result to the upstream peer or to stop the
27766 processing task in case of errors.
27768 2006-06-14 14:49:33 +0000 Tim-Philipp Müller <tim@centricular.net>
27770 gst/playback/gststreaminfo.c: Try GST_TAG_CODEC as fallback when extracting the codec name; more debug info.
27771 Original commit message from CVS:
27772 * gst/playback/gststreaminfo.c: (cb_probe):
27773 Try GST_TAG_CODEC as fallback when extracting the
27774 codec name; more debug info.
27776 2006-06-14 14:34:28 +0000 Tim-Philipp Müller <tim@centricular.net>
27778 ext/ogg/: Extract language tags from ogm subtitle streams, so that the subtitle menu choices are labelled correctly in
27779 Original commit message from CVS:
27780 * ext/ogg/Makefile.am:
27781 * ext/ogg/gstogmparse.c: (gst_ogm_parse_chain):
27782 Extract language tags from ogm subtitle streams, so that
27783 the subtitle menu choices are labelled correctly in
27784 Totem (fixes #344708).
27786 2006-06-14 09:13:54 +0000 Alessandro Decina <alessandro@nnva.org>
27788 ext/ogg/gstoggmux.c: Fix various leaks. Fixes #343699.
27789 Original commit message from CVS:
27790 Patch by: Alessandro Decina <alessandro at nnva dot org>
27791 * ext/ogg/gstoggmux.c: (gst_ogg_mux_get_type), (gst_ogg_mux_clear),
27792 (gst_ogg_mux_release_pad), (gst_ogg_mux_get_headers),
27793 (gst_ogg_mux_set_header_on_caps), (gst_ogg_mux_init_collectpads),
27794 (gst_ogg_mux_clear_collectpads), (gst_ogg_mux_change_state):
27795 Fix various leaks. Fixes #343699.
27796 Add x-smoke mime type.
27798 2006-06-14 08:17:45 +0000 Tim-Philipp Müller <tim@centricular.net>
27800 gst-libs/gst/riff/riff-ids.h: Add IDs for 'bext' chunks (see #343837).
27801 Original commit message from CVS:
27802 * gst-libs/gst/riff/riff-ids.h:
27803 Add IDs for 'bext' chunks (see #343837).
27805 2006-06-12 12:44:38 +0000 Young-Ho Cha <ganadist@chollian.net>
27807 gst/subparse/samiparse.c: Honour font face tags in SAMI subtitles (#344503).
27808 Original commit message from CVS:
27809 Patch by: Young-Ho Cha <ganadist at chollian net>
27810 * gst/subparse/samiparse.c: (sami_context_pop_state),
27811 (handle_start_font), (end_sami_element):
27812 Honour font face tags in SAMI subtitles (#344503).
27814 2006-06-11 20:41:56 +0000 Stefan Kost <ensonic@users.sourceforge.net>
27816 po/POTFILES.in: add missing files containing translatable strings
27817 Original commit message from CVS:
27819 add missing files containing translatable strings
27821 2006-06-11 19:55:32 +0000 Stefan Kost <ensonic@users.sourceforge.net>
27823 docs/libs/tmpl/.cvsignore: we don't want those *.sgml files in CVS either
27824 Original commit message from CVS:
27825 * docs/libs/tmpl/.cvsignore:
27826 we don't want those *.sgml files in CVS either
27828 2006-06-11 19:44:49 +0000 Stefan Kost <ensonic@users.sourceforge.net>
27831 Original commit message from CVS:
27832 * docs/libs/.cvsignore:
27833 * tests/check/elements/.cvsignore:
27834 * tests/check/libs/.cvsignore:
27837 2006-06-11 18:33:27 +0000 Stefan Kost <ensonic@users.sourceforge.net>
27839 docs/libs/Makefile.am: also commiting the changed Makefile.am (added more libs to the doc-build)
27840 Original commit message from CVS:
27841 * docs/libs/Makefile.am:
27842 also commiting the changed Makefile.am (added more libs to the
27845 2006-06-11 17:08:26 +0000 Stefan Kost <ensonic@users.sourceforge.net>
27847 docs/libs/: first batch of reordering things, add index & hierarchy
27848 Original commit message from CVS:
27849 * docs/libs/gst-plugins-base-libs-docs.sgml:
27850 * docs/libs/gst-plugins-base-libs-sections.txt:
27851 * docs/libs/gst-plugins-base-libs.types:
27852 first batch of reordering things, add index & hierarchy
27854 2006-06-11 14:08:54 +0000 Thomas Vander Stichele <thomas@apestaart.org>
27857 * ext/alsa/Makefile.am:
27858 * ext/cdparanoia/Makefile.am:
27859 * ext/gnomevfs/Makefile.am:
27860 * ext/libvisual/Makefile.am:
27861 * ext/ogg/Makefile.am:
27862 * ext/pango/Makefile.am:
27863 * ext/theora/Makefile.am:
27864 * ext/vorbis/Makefile.am:
27865 * sys/v4l/Makefile.am:
27866 * sys/ximage/Makefile.am:
27867 * sys/xvimage/Makefile.am:
27868 further clean up build
27869 Original commit message from CVS:
27870 further clean up build
27872 2006-06-11 12:14:18 +0000 Thomas Vander Stichele <thomas@apestaart.org>
27874 configure.ac: use GST_PKG_CHECK_MODULES, cleans up output
27875 Original commit message from CVS:
27877 use GST_PKG_CHECK_MODULES, cleans up output
27879 2006-06-11 12:10:55 +0000 Thomas Vander Stichele <thomas@apestaart.org>
27882 * win32/common/config.h:
27884 Original commit message from CVS:
27887 2006-06-10 18:52:03 +0000 Tim-Philipp Müller <tim@centricular.net>
27889 ext/gnomevfs/gstgnomevfsuri.c: Add support for burn:// URIs (#343385); const-ify things a bit, use G_N_ELEMENTS inste...
27890 Original commit message from CVS:
27891 * ext/gnomevfs/gstgnomevfsuri.c: (gst_gnomevfs_get_supported_uris):
27892 Add support for burn:// URIs (#343385); const-ify things a bit,
27893 use G_N_ELEMENTS instead of hard-coded array size.
27895 2006-06-10 18:25:07 +0000 Young-Ho Cha <ganadist@chollian.net>
27897 gst/subparse/samiparse.c: Fix up broken entities before passing them to libxml *sigh*. (#343303).
27898 Original commit message from CVS:
27899 Patch by: Young-Ho Cha <ganadist at chollian net>
27900 * gst/subparse/samiparse.c: (fix_invalid_entities), (parse_sami):
27901 Fix up broken entities before passing them to libxml *sigh*.
27904 2006-06-09 18:52:35 +0000 Thomas Vander Stichele <thomas@apestaart.org>
27909 Original commit message from CVS:
27912 === release 0.10.8 ===
27914 2006-06-09 18:49:26 +0000 Thomas Vander Stichele <thomas@apestaart.org>
27920 * docs/plugins/gst-plugins-base-plugins.args:
27921 * docs/plugins/inspect/plugin-adder.xml:
27922 * docs/plugins/inspect/plugin-alsa.xml:
27923 * docs/plugins/inspect/plugin-audioconvert.xml:
27924 * docs/plugins/inspect/plugin-audiorate.xml:
27925 * docs/plugins/inspect/plugin-audioresample.xml:
27926 * docs/plugins/inspect/plugin-audiotestsrc.xml:
27927 * docs/plugins/inspect/plugin-cdparanoia.xml:
27928 * docs/plugins/inspect/plugin-decodebin.xml:
27929 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
27930 * docs/plugins/inspect/plugin-gnomevfs.xml:
27931 * docs/plugins/inspect/plugin-libvisual.xml:
27932 * docs/plugins/inspect/plugin-ogg.xml:
27933 * docs/plugins/inspect/plugin-pango.xml:
27934 * docs/plugins/inspect/plugin-playbin.xml:
27935 * docs/plugins/inspect/plugin-subparse.xml:
27936 * docs/plugins/inspect/plugin-tcp.xml:
27937 * docs/plugins/inspect/plugin-theora.xml:
27938 * docs/plugins/inspect/plugin-typefindfunctions.xml:
27939 * docs/plugins/inspect/plugin-video4linux.xml:
27940 * docs/plugins/inspect/plugin-videorate.xml:
27941 * docs/plugins/inspect/plugin-videoscale.xml:
27942 * docs/plugins/inspect/plugin-videotestsrc.xml:
27943 * docs/plugins/inspect/plugin-volume.xml:
27944 * docs/plugins/inspect/plugin-vorbis.xml:
27945 * docs/plugins/inspect/plugin-ximagesink.xml:
27946 * docs/plugins/inspect/plugin-xvimagesink.xml:
27947 * win32/common/config.h:
27949 Original commit message from CVS:
27952 2006-06-07 11:04:09 +0000 Thomas Vander Stichele <thomas@apestaart.org>
27954 0.10.7.2 prerelease
27955 Original commit message from CVS:
27971 * win32/common/config.h:
27972 0.10.7.2 prerelease
27974 2006-06-07 11:03:03 +0000 Thomas Vander Stichele <thomas@apestaart.org>
27976 move last template doc snippets to source code and delete them
27977 Original commit message from CVS:
27978 * docs/libs/tmpl/gstaudio.sgml:
27979 * docs/libs/tmpl/gstcolorbalance.sgml:
27980 * docs/libs/tmpl/gstmixer.sgml:
27981 * docs/libs/tmpl/gstringbuffer.sgml:
27982 * docs/libs/tmpl/gsttuner.sgml:
27983 * docs/libs/tmpl/gstxoverlay.sgml:
27984 * gst-libs/gst/audio/audio.c:
27985 * gst-libs/gst/audio/gstringbuffer.c:
27986 * gst-libs/gst/interfaces/colorbalance.c:
27987 * gst-libs/gst/interfaces/mixer.c:
27988 * gst-libs/gst/interfaces/tuner.c:
27989 * gst-libs/gst/interfaces/xoverlay.c:
27990 move last template doc snippets to source code and delete them
27992 2006-06-06 16:26:59 +0000 Thomas Vander Stichele <thomas@apestaart.org>
27994 * gst/gdp/gstgdppay.c:
27996 Original commit message from CVS:
27999 2006-06-06 14:39:15 +0000 Thomas Vander Stichele <thomas@apestaart.org>
28001 configure.ac: enable building of GDP elements
28002 Original commit message from CVS:
28004 enable building of GDP elements
28005 * gst/gdp/gstgdpdepay.c: (gst_gdp_depay_chain):
28006 * gst/gdp/gstgdppay.c: (gst_gdp_pay_class_init),
28007 (gst_gdp_pay_init), (gst_gdp_buffer_from_caps),
28008 (gst_gdp_pay_buffer_from_buffer), (gst_gdp_buffer_from_event),
28009 (gst_gdp_pay_reset_streamheader), (gst_gdp_pay_sink_event),
28010 (gst_gdp_pay_set_property), (gst_gdp_pay_get_property),
28011 (gst_gdp_pay_change_state):
28012 * gst/gdp/gstgdppay.h:
28015 2006-06-06 11:13:18 +0000 Michael Smith <msmith@xiph.org>
28017 ext/theora/theoraparse.c: Mark DELTA_UNIT on non-keyframes.
28018 Original commit message from CVS:
28019 * ext/theora/theoraparse.c: (theora_parse_drain_queue_prematurely),
28020 (theora_parse_drain_queue):
28021 Mark DELTA_UNIT on non-keyframes.
28023 2006-06-03 21:06:49 +0000 Jan Schmidt <thaytan@mad.scientist.com>
28025 gst-libs/gst/audio/: Document better the fact that latency_time and buffer_time are values stored in microseconds, an...
28026 Original commit message from CVS:
28027 * gst-libs/gst/audio/gstbaseaudiosink.c:
28028 (gst_base_audio_sink_class_init), (gst_base_audio_sink_setcaps):
28029 * gst-libs/gst/audio/gstbaseaudiosink.h:
28030 * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_parse_caps),
28031 (gst_ring_buffer_samples_done):
28032 * gst-libs/gst/audio/gstringbuffer.h:
28033 Document better the fact that latency_time and buffer_time are values
28034 stored in microseconds, and not the usual GStreamer nanoseconds.
28035 Change the variables (compatibly) that store them from GstClockTime
28036 to guint64 to make it more clear that they're not storing clock times.
28037 Also, remove the bogus property description that says the user can
28038 specify -1 to get the default value, since that's never been the case.
28039 When computing the default segment size for the ring buffer, make it
28040 an integer number of samples.
28041 When the sub-class indicates a delay greater than the number of
28042 samples we've written return 0 from the audio sink get_time method.
28044 2006-06-02 17:01:02 +0000 Michael Smith <msmith@xiph.org>
28046 tests/check/: Don't busy-wait in tests; this was causing test timeouts very frequently when running under valgrind.
28047 Original commit message from CVS:
28048 * tests/check/elements/audioconvert.c: (set_channel_positions),
28049 (get_float_mc_caps), (get_int_mc_caps):
28050 * tests/check/elements/audioresample.c:
28051 * tests/check/elements/audiotestsrc.c: (GST_START_TEST):
28052 * tests/check/elements/videorate.c:
28053 * tests/check/elements/videotestsrc.c: (GST_START_TEST):
28054 * tests/check/elements/volume.c:
28055 * tests/check/elements/vorbisdec.c:
28056 * tests/check/pipelines/vorbisenc.c: (GST_START_TEST):
28057 Don't busy-wait in tests; this was causing test timeouts very
28058 frequently when running under valgrind.
28060 2006-06-02 16:45:59 +0000 Thomas Vander Stichele <thomas@apestaart.org>
28062 * gst/gdp/gstgdpdepay.c:
28063 * gst/gdp/gstgdppay.h:
28065 Original commit message from CVS:
28068 2006-06-02 16:35:12 +0000 Thomas Vander Stichele <thomas@apestaart.org>
28070 * tests/check/elements/multifdsink.c:
28071 fail_if_can_read is racy
28072 Original commit message from CVS:
28073 fail_if_can_read is racy
28075 2006-06-02 16:26:54 +0000 Thomas Vander Stichele <thomas@apestaart.org>
28077 gst/tcp/: make multifdsink properly deal with streamheader:
28078 Original commit message from CVS:
28080 * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_init),
28081 (gst_multi_fd_sink_remove_client_link),
28082 (gst_multi_fd_sink_client_queue_caps),
28083 (gst_multi_fd_sink_client_queue_buffer),
28084 (gst_multi_fd_sink_handle_client_write),
28085 (gst_multi_fd_sink_render):
28086 * gst/tcp/gstmultifdsink.h:
28087 make multifdsink properly deal with streamheader:
28088 - streamheader is taken from caps
28089 - buffers marked with IN_CAPS are not sent
28090 - streamheaders are sent, on connection, from the caps of the
28091 buffer where the client gets positioned to
28092 - further streamheader changes are done every time the client
28093 will receive a buffer with different caps
28094 * tests/check/elements/multifdsink.c: (GST_START_TEST),
28095 (gst_multifdsink_create_streamheader):
28098 2006-06-02 15:06:59 +0000 Michael Smith <msmith@xiph.org>
28100 ext/vorbis/vorbisdec.c: Reinstate limit on channel count. Vorbis does not define the meaning of > 6 channels, so they...
28101 Original commit message from CVS:
28102 * ext/vorbis/vorbisdec.c: (vorbis_handle_identification_packet):
28103 Reinstate limit on channel count. Vorbis does not define the meaning
28104 of > 6 channels, so they're just independent channels. Gstreamer
28105 currently has no mechanism to represent N independent channels.
28107 2006-06-02 14:23:34 +0000 Michael Smith <msmith@xiph.org>
28109 ext/vorbis/vorbisdec.c: Don't arbitrarily restrict channel counts and rate in vorbis.
28110 Original commit message from CVS:
28111 * ext/vorbis/vorbisdec.c: (vorbis_handle_identification_packet):
28112 Don't arbitrarily restrict channel counts and rate in vorbis.
28113 In terms of effects likely on real-world files, this fixes 96kHz
28114 playback of vorbis.
28116 2006-06-02 14:19:18 +0000 Michael Smith <msmith@xiph.org>
28118 gst/audioconvert/audioconvert.c: More correct float->int conversion.
28119 Original commit message from CVS:
28120 * gst/audioconvert/audioconvert.c: (float):
28121 More correct float->int conversion.
28123 2006-06-02 14:07:42 +0000 Michael Smith <msmith@xiph.org>
28125 ext/ogg/gstoggdemux.c: Don't accidently send GST_CLOCK_TIME_NONE as a new segment start value. Fixes g-critical on tr...
28126 Original commit message from CVS:
28127 * ext/ogg/gstoggdemux.c: (gst_ogg_demux_perform_seek):
28128 Don't accidently send GST_CLOCK_TIME_NONE as a new segment start
28129 value. Fixes g-critical on trying to play back ogg containing
28132 2006-06-02 10:34:12 +0000 Wim Taymans <wim.taymans@gmail.com>
28134 gst/playback/gstplaybasebin.*: Make the subtitle detection work from any thread so we don't deadlock. Fixes #343397.
28135 Original commit message from CVS:
28136 * gst/playback/gstplaybasebin.c: (group_create), (group_commit),
28138 * gst/playback/gstplaybasebin.h:
28139 Make the subtitle detection work from any thread so we don't
28140 deadlock. Fixes #343397.
28142 2006-06-02 10:28:32 +0000 Thomas Vander Stichele <thomas@apestaart.org>
28144 gst/gdp/gstgdppay.c: add crc-header and crc-payload properties don't error out on some things that are recoverable
28145 Original commit message from CVS:
28146 * gst/gdp/gstgdppay.c: (gst_gdp_pay_class_init),
28147 (gst_gdp_pay_init), (gst_gdp_buffer_from_caps),
28148 (gst_gdp_pay_buffer_from_buffer), (gst_gdp_buffer_from_event),
28149 (gst_gdp_pay_reset_streamheader), (gst_gdp_pay_chain),
28150 (gst_gdp_pay_sink_event), (gst_gdp_pay_set_property),
28151 (gst_gdp_pay_get_property):
28152 add crc-header and crc-payload properties
28153 don't error out on some things that are recoverable
28154 * tests/check/elements/gdppay.c: (GST_START_TEST), (gdppay_suite):
28157 2006-06-02 09:17:07 +0000 Thomas Vander Stichele <thomas@apestaart.org>
28159 * gst/tcp/gsttcp.c:
28160 show type number when packet is of the wrong type
28161 Original commit message from CVS:
28162 show type number when packet is of the wrong type
28164 2006-06-01 23:04:32 +0000 Thomas Vander Stichele <thomas@apestaart.org>
28166 gst/volume/Makefile.am: Seriously, it's not *that* hard to get compilation right. Even a drunk can do it ! Add LIBOI...
28167 Original commit message from CVS:
28168 * gst/volume/Makefile.am:
28169 Seriously, it's not *that* hard to get compilation right. Even
28170 a drunk can do it ! Add LIBOIL CFLAGS and LIBS
28172 2006-06-01 22:00:23 +0000 Stefan Kost <ensonic@users.sourceforge.net>
28174 Fix more gobject macros: obj<->klass, GstXXX<->GstXXXClass
28175 Original commit message from CVS:
28176 * ext/alsaspdif/alsaspdifsink.h:
28177 * ext/amrwb/gstamrwbdec.h:
28178 * ext/amrwb/gstamrwbenc.h:
28179 * ext/amrwb/gstamrwbparse.h:
28180 * ext/arts/gst_arts.h:
28181 * ext/artsd/gstartsdsink.h:
28182 * ext/audiofile/gstafparse.h:
28183 * ext/audiofile/gstafsink.h:
28184 * ext/audiofile/gstafsrc.h:
28185 * ext/audioresample/gstaudioresample.h:
28186 * ext/bz2/gstbz2dec.h:
28187 * ext/bz2/gstbz2enc.h:
28188 * ext/dirac/gstdiracdec.h:
28189 * ext/directfb/dfbvideosink.h:
28190 * ext/divx/gstdivxdec.h:
28191 * ext/divx/gstdivxenc.h:
28192 * ext/dts/gstdtsdec.h:
28193 * ext/faac/gstfaac.h:
28194 * ext/gsm/gstgsmdec.h:
28195 * ext/gsm/gstgsmenc.h:
28196 * ext/ivorbis/vorbisenc.h:
28197 * ext/libfame/gstlibfame.h:
28198 * ext/nas/nassink.h:
28199 * ext/neon/gstneonhttpsrc.h:
28200 * ext/polyp/polypsink.h:
28201 * ext/sdl/sdlaudiosink.h:
28202 * ext/sdl/sdlvideosink.h:
28203 * ext/shout/gstshout.h:
28204 * ext/snapshot/gstsnapshot.h:
28205 * ext/sndfile/gstsf.h:
28206 * ext/swfdec/gstswfdec.h:
28207 * ext/tarkin/gsttarkindec.h:
28208 * ext/tarkin/gsttarkinenc.h:
28209 * ext/theora/theoradec.h:
28210 * ext/wavpack/gstwavpackdec.h:
28211 * ext/wavpack/gstwavpackparse.h:
28212 * ext/xine/gstxine.h:
28213 * ext/xvid/gstxviddec.h:
28214 * ext/xvid/gstxvidenc.h:
28215 * gst/cdxaparse/gstcdxaparse.h:
28216 * gst/cdxaparse/gstcdxastrip.h:
28217 * gst/colorspace/gstcolorspace.h:
28218 * gst/festival/gstfestival.h:
28219 * gst/freeze/gstfreeze.h:
28220 * gst/gdp/gstgdpdepay.h:
28221 * gst/gdp/gstgdppay.h:
28222 * gst/modplug/gstmodplug.h:
28223 * gst/mpeg1sys/gstmpeg1systemencode.h:
28224 * gst/mpeg1videoparse/gstmp1videoparse.h:
28225 * gst/mpeg2sub/gstmpeg2subt.h:
28226 * gst/mpegaudioparse/gstmpegaudioparse.h:
28227 * gst/multifilesink/gstmultifilesink.h:
28228 * gst/overlay/gstoverlay.h:
28229 * gst/playondemand/gstplayondemand.h:
28230 * gst/qtdemux/qtdemux.h:
28231 * gst/rtjpeg/gstrtjpegdec.h:
28232 * gst/rtjpeg/gstrtjpegenc.h:
28233 * gst/smooth/gstsmooth.h:
28234 * gst/smoothwave/gstsmoothwave.h:
28235 * gst/spectrum/gstspectrum.h:
28236 * gst/speed/gstspeed.h:
28237 * gst/stereo/gststereo.h:
28238 * gst/switch/gstswitch.h:
28239 * gst/tta/gstttadec.h:
28240 * gst/tta/gstttaparse.h:
28241 * gst/videodrop/gstvideodrop.h:
28242 * gst/xingheader/gstxingmux.h:
28243 * sys/directdraw/gstdirectdrawsink.h:
28244 * sys/directsound/gstdirectsoundsink.h:
28245 * sys/dxr3/dxr3audiosink.h:
28246 * sys/dxr3/dxr3spusink.h:
28247 * sys/dxr3/dxr3videosink.h:
28248 * sys/qcam/gstqcamsrc.h:
28249 * sys/vcd/vcdsrc.h:
28250 Fix more gobject macros: obj<->klass, GstXXX<->GstXXXClass
28252 2006-06-01 20:39:30 +0000 Stefan Kost <ensonic@users.sourceforge.net>
28254 gst/volume/gstvolume.*: rewrite the passthrough check, split _int16 and _int16_clamp, fix another property desc., rem...
28255 Original commit message from CVS:
28256 * gst/volume/gstvolume.c: (volume_choose_func),
28257 (volume_update_real_volume), (gst_volume_class_init),
28258 (gst_volume_init), (volume_process_float), (volume_process_int16),
28259 (volume_process_int16_clamp), (volume_set_caps),
28260 (volume_transform_ip), (plugin_init):
28261 * gst/volume/gstvolume.h:
28262 rewrite the passthrough check, split _int16 and _int16_clamp, fix
28263 another property desc., remove unused param from process function
28264 * tests/check/elements/volume.c: (volume_suite):
28265 reactivate the passthrough test
28267 2006-06-01 19:19:51 +0000 Stefan Kost <ensonic@users.sourceforge.net>
28269 Fix more gobject macros: obj<->klass, GstXXX<->GstXXXClass
28270 Original commit message from CVS:
28271 * ext/alsa/gstalsamixerelement.h:
28272 * ext/alsa/gstalsamixeroptions.h:
28273 * ext/alsa/gstalsamixertrack.h:
28274 * ext/gnomevfs/gstgnomevfssink.h:
28275 * ext/gnomevfs/gstgnomevfssrc.h:
28276 * ext/theora/gsttheoradec.h:
28277 * ext/theora/gsttheoraenc.h:
28278 * ext/theora/gsttheoraparse.h:
28279 * ext/vorbis/vorbisparse.h:
28280 * gst-libs/gst/audio/gstaudioclock.h:
28281 * gst-libs/gst/audio/gstaudiofilter.h:
28282 * gst-libs/gst/rtp/gstbasertpaudiopayload.h:
28283 * gst/audioconvert/gstaudioconvert.h:
28284 * gst/audioresample/gstaudioresample.h:
28285 * gst/audiotestsrc/gstaudiotestsrc.h:
28286 * gst/ffmpegcolorspace/gstffmpegcolorspace.h:
28287 * gst/playback/gststreamselector.h:
28288 * gst/tcp/gstmultifdsink.h:
28289 * gst/tcp/gsttcpclientsink.h:
28290 * gst/tcp/gsttcpclientsrc.h:
28291 * gst/tcp/gsttcpserversink.h:
28292 * gst/tcp/gsttcpserversrc.h:
28293 * gst/videorate/gstvideorate.h:
28294 * gst/videoscale/gstvideoscale.h:
28295 * gst/videotestsrc/gstvideotestsrc.h:
28296 * gst/volume/gstvolume.h:
28297 * sys/v4l/gstv4ljpegsrc.h:
28298 * sys/v4l/gstv4lmjpegsink.h:
28299 * sys/v4l/gstv4lmjpegsrc.h:
28300 * sys/v4l/gstv4lsrc.h:
28301 * sys/ximage/ximagesink.h:
28302 * sys/xvimage/xvimagesink.h:
28303 * tests/old/testsuite/alsa/sinesrc.h:
28304 Fix more gobject macros: obj<->klass, GstXXX<->GstXXXClass
28306 2006-05-31 16:56:45 +0000 Thomas Vander Stichele <thomas@apestaart.org>
28308 * tests/check/elements/multifdsink.c:
28309 remove wrong commit
28310 Original commit message from CVS:
28311 remove wrong commit
28313 2006-05-31 16:21:48 +0000 Wim Taymans <wim.taymans@gmail.com>
28315 ext/libvisual/visual.c: Handle DISCONT.
28316 Original commit message from CVS:
28317 * ext/libvisual/visual.c: (gst_visual_reset),
28318 (gst_visual_sink_setcaps), (gst_visual_sink_event),
28319 (gst_visual_src_event), (get_buffer), (gst_visual_chain):
28321 Use running time before doing QoS.
28324 2006-05-31 14:17:30 +0000 Thomas Vander Stichele <thomas@apestaart.org>
28326 docs/libs/Makefile.am: set a magic variable to indicate we know the docs are incomplete
28327 Original commit message from CVS:
28328 * docs/libs/Makefile.am:
28329 set a magic variable to indicate we know the docs are incomplete
28331 2006-05-30 20:33:59 +0000 Sébastien Moutte <sebastien@moutte.net>
28333 win32/common/libgstvideo.def: export gst_video_calculate_display_ratio
28334 Original commit message from CVS:
28335 * win32/common/libgstvideo.def:
28336 export gst_video_calculate_display_ratio
28337 * win32/vs6/libgstvideoscale.dsp:
28338 add link to libgstvideo-0.10.lib
28340 2006-05-30 19:00:39 +0000 Tim-Philipp Müller <tim@centricular.net>
28342 gst/playback/gstplaybasebin.c: Throw a more comprehensible error for rtsp:// URIs (rather than erroring out with a ne...
28343 Original commit message from CVS:
28344 * gst/playback/gstplaybasebin.c: (gen_source_element):
28345 Throw a more comprehensible error for rtsp:// URIs (rather
28346 than erroring out with a negotiation error later on) until
28347 we fix playbin to handle rtspsrc etc.
28349 2006-05-30 16:09:36 +0000 Wim Taymans <wim.taymans@gmail.com>
28351 ext/pango/gsttextoverlay.c: Added some FIXMEs.
28352 Original commit message from CVS:
28353 * ext/pango/gsttextoverlay.c: (gst_text_overlay_src_event),
28354 (gst_text_overlay_text_event):
28357 2006-05-30 16:07:50 +0000 Wim Taymans <wim.taymans@gmail.com>
28359 gst/adder/gstadder.*: Implement release_request_pad.
28360 Original commit message from CVS:
28361 * gst/adder/gstadder.c: (gst_adder_class_init), (gst_adder_init),
28362 (gst_adder_request_new_pad), (gst_adder_release_pad):
28363 * gst/adder/gstadder.h:
28364 Implement release_request_pad.
28365 Make padcounter atomic.
28366 * tests/check/elements/adder.c: (GST_START_TEST), (adder_suite):
28367 Added check for release_pad in adder.
28369 2006-05-30 16:04:14 +0000 Wim Taymans <wim.taymans@gmail.com>
28371 ext/ogg/gstoggdemux.c: Fix build again.
28372 Original commit message from CVS:
28373 * ext/ogg/gstoggdemux.c: (gst_ogg_chain_new_stream):
28376 2006-05-30 14:59:24 +0000 Thomas Vander Stichele <thomas@apestaart.org>
28378 ext/ogg/gstoggdemux.c: add more debugging clean up printf formats for granulepos and serialno
28379 Original commit message from CVS:
28380 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_typefind),
28381 (gst_ogg_demux_chain_elem_pad), (gst_ogg_demux_queue_data),
28382 (gst_ogg_demux_chain_peer), (gst_ogg_pad_submit_packet),
28383 (gst_ogg_pad_submit_page), (gst_ogg_chain_new_stream),
28384 (gst_ogg_demux_seek), (gst_ogg_demux_get_data),
28385 (gst_ogg_demux_get_next_page), (gst_ogg_demux_do_seek),
28386 (gst_ogg_demux_bisect_forward_serialno),
28387 (gst_ogg_demux_read_chain), (gst_ogg_demux_find_chains),
28388 (gst_ogg_demux_chain), (gst_ogg_demux_loop), (gst_ogg_print):
28390 clean up printf formats for granulepos and serialno
28392 2006-05-30 14:31:43 +0000 Thomas Vander Stichele <thomas@apestaart.org>
28394 * tests/check/elements/multifdsink.c:
28395 * tests/check/generic/states.c:
28396 properly fail if we can't make an element
28397 Original commit message from CVS:
28398 properly fail if we can't make an element
28400 2006-05-30 13:22:58 +0000 Michael Smith <msmith@xiph.org>
28402 ext/vorbis/vorbisenc.*: Multi-channel caps negotiation, so we can do proper multichannel vorbis encoding, negotiated ...
28403 Original commit message from CVS:
28404 * ext/vorbis/vorbisenc.c: (raw_caps_factory),
28405 (gst_vorbis_enc_class_init), (gst_vorbis_enc_dispose),
28406 (gst_vorbis_enc_generate_sink_caps), (gst_vorbis_enc_sink_getcaps),
28407 (gst_vorbis_enc_init), (gst_vorbis_enc_buffer_from_header_packet),
28408 (gst_vorbis_enc_chain), (gst_vorbis_enc_change_state):
28409 * ext/vorbis/vorbisenc.h:
28410 Multi-channel caps negotiation, so we can do proper multichannel
28411 vorbis encoding, negotiated through audioconvert.
28413 2006-05-30 11:45:52 +0000 Wim Taymans <wim.taymans@gmail.com>
28415 tests/check/elements/adder.c: Added check to show that #339935 is fixed with ongoing adder and collectpads fixes.
28416 Original commit message from CVS:
28417 * tests/check/elements/adder.c: (test_event_message_received),
28418 (test_play_twice_message_received), (GST_START_TEST),
28420 Added check to show that #339935 is fixed with ongoing
28421 adder and collectpads fixes.
28423 2006-05-29 17:19:48 +0000 Wim Taymans <wim.taymans@gmail.com>
28425 gst/adder/gstadder.c: Don't leak pad name.
28426 Original commit message from CVS:
28427 * gst/adder/gstadder.c: (gst_adder_request_new_pad):
28428 Don't leak pad name.
28430 2006-05-29 15:49:53 +0000 Wim Taymans <wim.taymans@gmail.com>
28432 gst/adder/gstadder.c: Fix adder seeking.
28433 Original commit message from CVS:
28434 * gst/adder/gstadder.c: (gst_adder_query_duration),
28435 (forward_event_func), (forward_event), (gst_adder_src_event):
28437 Make query/seeking code threadsafe.
28438 * tests/check/Makefile.am:
28439 * tests/check/elements/adder.c: (test_event_message_received),
28440 (GST_START_TEST), (test_play_twice_message_received):
28441 Fix adder test case.
28443 2006-05-29 13:21:00 +0000 Young-Ho Cha <ganadist@chollian.net>
28445 gst/playback/gstplaybasebin.*: Add 'subtitle-encoding' property to playbin, so applications can force a subtitle enco...
28446 Original commit message from CVS:
28447 Patch by: Young-Ho Cha <ganadist at chollian net>
28448 * gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init),
28449 (gst_play_base_bin_init), (gst_play_base_bin_dispose),
28450 (set_encoding_element), (decodebin_element_added_cb),
28451 (decodebin_element_removed_cb), (setup_subtitle), (setup_source),
28452 (gst_play_base_bin_set_property), (gst_play_base_bin_get_property):
28453 * gst/playback/gstplaybasebin.h:
28454 Add 'subtitle-encoding' property to playbin, so applications can
28455 force a subtitle encoding for non-UTF8 subtitles (#342268).
28456 * gst/subparse/gstsubparse.c: (gst_sub_parse_class_init),
28457 (gst_sub_parse_set_property):
28458 Rename recently-added 'encoding' property to 'subtitle-encoding'
28459 (so it can be proxied by playbin/decodebin in a generic way
28460 with less danger of false positives).
28462 2006-05-29 11:04:48 +0000 Michael Smith <msmith@xiph.org>
28464 gst/audioconvert/gstaudioconvert.c: Patch from #341562: give more specific audio caps in get_caps, so that basetransf...
28465 Original commit message from CVS:
28466 * gst/audioconvert/gstaudioconvert.c: (make_lossless_changes),
28467 (append_with_other_format), (set_structure_widths),
28468 (gst_audio_convert_transform_caps):
28469 Patch from #341562: give more specific audio caps in get_caps, so
28470 that basetransform can make better decisions on what caps to
28473 2006-05-28 20:04:12 +0000 Stefan Kost <ensonic@users.sourceforge.net>
28475 tests/check/elements/volume.c: make it compile again
28476 Original commit message from CVS:
28477 * tests/check/elements/volume.c:
28478 make it compile again
28480 2006-05-28 19:56:25 +0000 Stefan Kost <ensonic@users.sourceforge.net>
28482 tests/check/elements/volume.c: disable test until #343196 gets resolved
28483 Original commit message from CVS:
28484 * tests/check/elements/volume.c: (volume_suite):
28485 disable test until #343196 gets resolved
28487 2006-05-28 19:42:27 +0000 Stefan Kost <ensonic@users.sourceforge.net>
28489 gst/adder/gstadder.c: Make it easier to copy&paste
28490 Original commit message from CVS:
28491 * gst/adder/gstadder.c: (gst_adder_get_type):
28492 Make it easier to copy&paste
28493 * gst/volume/Makefile.am:
28494 * gst/volume/gstvolume.c: (volume_update_real_volume),
28495 (gst_volume_set_volume), (gst_volume_set_mute),
28496 (gst_volume_class_init), (volume_process_int16), (volume_set_caps),
28497 (volume_transform_ip), (volume_update_mute),
28498 (volume_update_volume):
28499 * gst/volume/gstvolume.h:
28500 Add own debug category, move duplicate code to helper function, fix
28501 property texts, add more comments and prepare ffor liboil-goodness
28502 * tests/check/Makefile.am:
28503 * tests/check/elements/volume.c: (GST_START_TEST), (volume_suite):
28504 add test for mute and passtrough case, be a bit more verbose to track
28506 * tests/check/generic/states.c: (GST_START_TEST):
28507 catch elements that fail to instantiate
28509 2006-05-28 09:37:18 +0000 Edward Hervey <bilboed@bilboed.com>
28511 tests/check/pipelines/: Comment out tests using parse_launch() if core was built without parsing capabilities.
28512 Original commit message from CVS:
28513 * tests/check/pipelines/simple-launch-lines.c:
28514 * tests/check/pipelines/theoraenc.c:
28515 * tests/check/pipelines/vorbisenc.c:
28516 Comment out tests using parse_launch() if core was built without
28517 parsing capabilities.
28519 2006-05-27 13:34:03 +0000 Edward Hervey <bilboed@bilboed.com>
28521 tests/check/Makefile.am: Extra bonus points for whoever explains to ensonic that you are meant to test unit tests tho...
28522 Original commit message from CVS:
28523 * tests/check/Makefile.am:
28524 Extra bonus points for whoever explains to ensonic that you are meant
28525 to test unit tests thoroughly before commiting them, especially if
28526 you know it's going to break.
28527 De-activated element/adder tests.
28529 2006-05-27 13:09:16 +0000 Edward Hervey <bilboed@bilboed.com>
28531 gst/ffmpegcolorspace/gstffmpegcodecmap.c: Marking caps conversion issues as GST_WARNING is way too verbose,
28532 Original commit message from CVS:
28533 * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
28534 (gst_ffmpeg_pixfmt_to_caps), (gst_ffmpeg_smpfmt_to_caps):
28535 Marking caps conversion issues as GST_WARNING is way too verbose,
28536 Moving them to GST_LOG.
28538 2006-05-27 11:26:18 +0000 Tim-Philipp Müller <tim@centricular.net>
28540 README: Replace current README (containing the release notes from some 0.9.x version) with a proper README taken from...
28541 Original commit message from CVS:
28543 Replace current README (containing the release notes from
28544 some 0.9.x version) with a proper README taken from the core.
28546 2006-05-26 15:52:23 +0000 Wim Taymans <wim.taymans@gmail.com>
28548 ext/vorbis/vorbisdec.c: Small cleanups.
28549 Original commit message from CVS:
28550 * ext/vorbis/vorbisdec.c: (vorbis_dec_src_query),
28551 (vorbis_dec_src_event), (vorbis_dec_sink_event), (vorbis_do_clip),
28552 (vorbis_dec_push), (vorbis_handle_data_packet), (vorbis_dec_chain),
28553 (vorbis_dec_change_state):
28556 Clip output samples to segment boundaries.
28558 2006-05-26 11:17:57 +0000 Jan Schmidt <thaytan@mad.scientist.com>
28560 sys/ximage/ximagesink.c: Improve the errors produced on bad output, including some human readable description strings.
28561 Original commit message from CVS:
28562 * sys/ximage/ximagesink.c: (gst_ximagesink_ximage_new),
28563 (gst_ximagesink_xcontext_get), (gst_ximagesink_show_frame):
28564 Improve the errors produced on bad output, including some human
28565 readable description strings.
28566 Handle the (theoretical for ximagesink) case where the XServer
28567 has a different idea about the size required for a particular
28568 frame and gives us too small a memory allocation.
28570 2006-05-26 10:18:41 +0000 Jan Schmidt <thaytan@mad.scientist.com>
28573 Mention bugs fixed by previous commit
28574 Original commit message from CVS:
28575 Mention bugs fixed by previous commit
28577 2006-05-26 09:40:35 +0000 Jan Schmidt <thaytan@mad.scientist.com>
28579 sys/xvimage/xvimagesink.c: Improve the errors produced on bad output, including some human readable description strings.
28580 Original commit message from CVS:
28581 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_new),
28582 (gst_xvimagesink_get_xv_support), (gst_xvimagesink_xcontext_get),
28583 (gst_xvimagesink_get_format_from_caps), (gst_xvimagesink_setcaps),
28584 (gst_xvimagesink_show_frame), (gst_xvimagesink_buffer_alloc):
28585 Improve the errors produced on bad output, including some human
28586 readable description strings.
28587 Handle RGB Xv formats properly by transforming them into our
28588 big-endian caps description.
28589 Use gst_caps_truncate to ensure that we never try and choose a
28590 non-fixed caps in buffer_alloc.
28591 Handle the case where the XServer has a different idea about the size
28592 required for a particular frame and gives us too small a memory
28594 Use -1 to indicate 'no image format', because 0 is a valid XServer
28595 image format number.
28596 Put RGB Xv formats at the end of the caps, so that we always prefer
28598 Iterate the available Xv Encodings to determine the maximum width and
28599 height, and then return that in our caps.
28601 2006-05-25 16:50:17 +0000 Jan Schmidt <thaytan@mad.scientist.com>
28603 gst/playback/gstdecodebin.c: When there is only one unfinished pad and it receives an event that doesn't match our re...
28604 Original commit message from CVS:
28605 * gst/playback/gstdecodebin.c: (remove_fakesink), (pad_probe):
28606 When there is only one unfinished pad and it receives an event that
28607 doesn't match our requirements, we need to set alldone=FALSE so that
28608 the fakesink is not removed yet.
28610 2006-05-25 09:32:31 +0000 Tim-Philipp Müller <tim@centricular.net>
28612 ext/ogg/gstoggdemux.c: Use gst_type_find_helper_for_buffer() to find the type of stream from the first packet.
28613 Original commit message from CVS:
28614 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_typefind):
28615 Use gst_type_find_helper_for_buffer() to find the type
28616 of stream from the first packet.
28618 Bump requirements to core CVS (needed for vorbis
28619 typefinding to work).
28621 2006-05-24 08:34:53 +0000 Edward Hervey <bilboed@bilboed.com>
28623 gst/typefind/gsttypefindfunctions.c: Added the 'prfl' atom type which MQV (no, it's not a typo) files contain.
28624 Original commit message from CVS:
28625 * gst/typefind/gsttypefindfunctions.c: (qt_type_find):
28626 Added the 'prfl' atom type which MQV (no, it's not a typo) files contain.
28627 Else they play perfectly fine with qtdemux.
28629 2006-05-23 20:38:56 +0000 Stefan Kost <ensonic@users.sourceforge.net>
28631 make more debug catagories static
28632 Original commit message from CVS:
28633 * ext/theora/theoradec.c:
28634 * ext/theora/theoraenc.c:
28635 * ext/theora/theoraparse.c: (gst_theora_parse_class_init):
28636 * gst/audiorate/gstaudiorate.c:
28637 make more debug catagories static
28638 * tests/check/Makefile.am:
28639 * tests/check/elements/adder.c: (message_received),
28640 (test_event_message_received), (GST_START_TEST),
28641 (test_play_twice_message_received), (adder_suite):
28642 added test case for using element twice, extra bonus points for anyone
28643 who can make these test run reliably
28645 2006-05-23 15:18:40 +0000 Tim-Philipp Müller <tim@centricular.net>
28647 ext/theora/theoradec.c: Make work with time-stamped input buffers that do not have a granulepos in BUFFER_OFFSET_END ...
28648 Original commit message from CVS:
28649 * ext/theora/theoradec.c: (theora_dec_chain):
28650 Make work with time-stamped input buffers that do not
28651 have a granulepos in BUFFER_OFFSET_END (like theora
28652 buffers coming from matroskademux). Fixes #342448.
28654 2006-05-22 15:53:07 +0000 Thomas Vander Stichele <thomas@apestaart.org>
28656 gst/gdp/: Handle error cases when calling functions do downwards state change after parent's change_state
28657 Original commit message from CVS:
28658 * gst/gdp/gstgdpdepay.c: (gst_gdp_depay_chain),
28659 (gst_gdp_depay_change_state):
28660 * gst/gdp/gstgdpdepay.h:
28661 * gst/gdp/gstgdppay.c: (gst_gdp_pay_reset_streamheader),
28662 (gst_gdp_pay_chain), (gst_gdp_pay_sink_event),
28663 (gst_gdp_pay_change_state):
28664 * gst/gdp/gstgdppay.h:
28665 Handle error cases when calling functions
28666 do downwards state change after parent's change_state
28667 * tests/check/elements/gdpdepay.c: (GST_START_TEST):
28668 * tests/check/elements/gdppay.c: (GST_START_TEST):
28671 2006-05-22 13:25:52 +0000 Thomas Vander Stichele <thomas@apestaart.org>
28673 adding GDP payloader and depayloader. Build integration will follow later when the GDP issues for core are sorted out.
28674 Original commit message from CVS:
28675 * gst/gdp/Makefile.am:
28676 * gst/gdp/gstgdp.c: (plugin_init):
28677 * gst/gdp/gstgdpdepay.c: (gst_gdp_depay_base_init),
28678 (gst_gdp_depay_class_init), (gst_gdp_depay_init),
28679 (gst_gdp_depay_finalize), (gst_gdp_depay_chain),
28680 (gst_gdp_depay_change_state), (gst_gdp_depay_plugin_init):
28681 * gst/gdp/gstgdpdepay.h:
28682 * gst/gdp/gstgdppay.c: (gst_gdp_pay_base_init),
28683 (gst_gdp_pay_class_init), (gst_gdp_pay_init),
28684 (gst_gdp_pay_dispose), (gst_gdp_stamp_buffer),
28685 (gst_gdp_buffer_from_caps), (gst_gdp_pay_buffer_from_buffer),
28686 (gst_gdp_buffer_from_event), (gst_gdp_pay_reset_streamheader),
28687 (gst_gdp_queue_buffer), (gst_gdp_pay_chain),
28688 (gst_gdp_pay_sink_event), (gst_gdp_pay_change_state),
28689 (gst_gdp_pay_plugin_init):
28690 * gst/gdp/gstgdppay.h:
28691 * tests/check/Makefile.am:
28692 * tests/check/elements/gdpdepay.c: (setup_gdpdepay),
28693 (cleanup_gdpdepay), (gdpdepay_push_per_byte), (GST_START_TEST),
28694 (setup_gdpdepay_streamheader), (gdpdepay_suite), (main):
28695 * tests/check/elements/gdppay.c: (setup_gdppay), (cleanup_gdppay),
28696 (GST_START_TEST), (setup_gdppay_streamheader), (gdppay_suite),
28698 adding GDP payloader and depayloader. Build integration will
28699 follow later when the GDP issues for core are sorted out.
28701 2006-05-22 11:42:03 +0000 Peter Kjellerstedt <pkj@axis.com>
28703 gst/tcp/Makefile.am: fdstresstest doesn't need Gtk+, fix compilation if gtk is not available (#342566).
28704 Original commit message from CVS:
28705 Patch by: Peter Kjellerstedt <pkj at axis com>
28706 * gst/tcp/Makefile.am:
28707 fdstresstest doesn't need Gtk+, fix compilation if
28708 gtk is not available (#342566).
28710 2006-05-19 17:57:56 +0000 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
28712 gst-libs/gst/rtp/gstbasertpaudiopayload.c: 80 line columns
28713 Original commit message from CVS:
28714 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
28716 Removed redundant floor()
28718 2006-05-19 15:00:43 +0000 Tim-Philipp Müller <tim@centricular.net>
28720 gst-libs/gst/riff/riff-read.c: On second thought, just skip JUNK chunks automatically, so the caller doesn't have to ...
28721 Original commit message from CVS:
28722 * gst-libs/gst/riff/riff-read.c: (gst_riff_read_chunk):
28723 On second thought, just skip JUNK chunks automatically, so
28724 the caller doesn't have to handle this. Fixes #342345.
28725 Also, return GST_FLOW_UNEXPECTED if we get a short read,
28726 not GST_FLOW_ERROR.
28728 2006-05-19 13:37:55 +0000 Tim-Philipp Müller <tim@centricular.net>
28730 gst-libs/gst/riff/riff-read.c: Don't bail out on JUNK chunks with a size of 0 (would try to pull_range 0 bytes before...
28731 Original commit message from CVS:
28732 * gst-libs/gst/riff/riff-read.c: (gst_riff_read_chunk):
28733 Don't bail out on JUNK chunks with a size of 0 (would try to
28734 pull_range 0 bytes before, which sources don't like too much).
28737 2006-05-19 13:02:46 +0000 Jan Schmidt <thaytan@mad.scientist.com>
28739 Use the gstutil scaling function to preserve 64 bits while calculating output width and height from the display-aspec...
28740 Original commit message from CVS:
28741 * gst/videoscale/gstvideoscale.c: (gst_video_scale_fixate_caps):
28742 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_setcaps):
28743 Use the gstutil scaling function to preserve 64 bits while calculating
28744 output width and height from the display-aspect-ratio. (A continuation
28747 2006-05-19 11:50:17 +0000 Jan Schmidt <thaytan@mad.scientist.com>
28749 sys/xvimage/xvimagesink.*: When performing buffer allocations, remember the caps and image format we return so that i...
28750 Original commit message from CVS:
28751 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xcontext_clear),
28752 (gst_xvimagesink_buffer_alloc):
28753 * sys/xvimage/xvimagesink.h:
28754 When performing buffer allocations, remember the caps and image format
28755 we return so that if the same caps are asked for next time we can
28756 return them immediately without doing any caps intersections.
28758 2006-05-18 23:00:02 +0000 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
28760 gst-libs/gst/rtp/README: Some new documentation
28761 Original commit message from CVS:
28762 2006-05-18 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
28763 * gst-libs/gst/rtp/README:
28764 Some new documentation
28765 * gst-libs/gst/rtp/gstrtpbuffer.h:
28766 Added GST_RTP_PAYLOAD_DYNAMIC_STRING for use by children
28767 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
28768 * gst-libs/gst/rtp/gstbasertpaudiopayload.h:
28769 New RTP audio base payloader class. Supports frame or sample based codecs.
28770 Not enabled in Makefile.am until approved.
28772 2006-05-18 20:30:26 +0000 Tim-Philipp Müller <tim@centricular.net>
28774 tests/check/elements/alsa.c: Fix test case: don't try to free NULL GValueArray when there are no devices.
28775 Original commit message from CVS:
28776 * tests/check/elements/alsa.c: (test_device_property_probe):
28777 Fix test case: don't try to free NULL GValueArray when there
28780 2006-05-18 19:21:53 +0000 Tim-Philipp Müller <tim@centricular.net>
28782 tests/check/: Add simple test that runs a device property probe on alsasrc, alsasink and alsamixer. Disable valgrind ...
28783 Original commit message from CVS:
28784 * tests/check/Makefile.am:
28785 * tests/check/elements/alsa.c: (test_device_property_probe),
28786 (alsa_suite), (main):
28787 Add simple test that runs a device property probe on alsasrc,
28788 alsasink and alsamixer. Disable valgrind check for now (too
28789 many leaks in libasound, and valgrind ignored my suppressions
28792 2006-05-18 17:19:39 +0000 Martin Szulecki <gnomebugzilla@sukimashita.com>
28794 ext/alsa/: Clean up and simplify alsa device probing. Make it actually work for multiple classes. Don't cache results...
28795 Original commit message from CVS:
28796 * ext/alsa/gstalsadeviceprobe.c: (gst_alsa_get_device_list),
28797 (gst_alsa_device_property_probe_probe_property),
28798 (gst_alsa_device_property_probe_needs_probe),
28799 (gst_alsa_device_property_probe_get_values),
28800 (gst_alsa_type_add_device_property_probe_interface):
28801 * ext/alsa/gstalsadeviceprobe.h:
28802 * ext/alsa/gstalsamixerelement.c:
28803 (gst_alsa_mixer_element_init_interfaces):
28804 * ext/alsa/gstalsamixerelement.h:
28805 Clean up and simplify alsa device probing. Make it actually work
28806 for multiple classes. Don't cache results any longer.
28807 * ext/alsa/gstalsasink.c: (gst_alsasink_init_interfaces),
28808 (gst_alsasink_init):
28809 * ext/alsa/gstalsasrc.c: (gst_alsasrc_dispose),
28810 (gst_alsasrc_interface_supported), (gst_implements_interface_init),
28811 (gst_alsasrc_init_interfaces), (gst_alsasrc_set_property):
28812 Make alsasink and alsasrc implement the GstPropertyProbe interface
28813 for device probing (#342181).
28814 Patch by: Martin Szulecki <gnomebugzilla at sukimashita com>
28816 2006-05-18 10:05:23 +0000 Tim-Philipp Müller <tim@centricular.net>
28818 gst/subparse/samiparse.c: Don't ignore return value of strtol (++compiler_happiness).
28819 Original commit message from CVS:
28820 * gst/subparse/samiparse.c: (handle_start_font):
28821 Don't ignore return value of strtol (++compiler_happiness).
28823 2006-05-17 17:49:10 +0000 Young-Ho Cha <ganadist@chollian.net>
28825 gst/subparse/gstsubparse.*: Add 'encoding' property (#341681).
28826 Original commit message from CVS:
28827 Patch by: Young-Ho Cha <ganadist chollian net>
28828 * gst/subparse/gstsubparse.c: (gst_sub_parse_dispose),
28829 (gst_sub_parse_class_init), (gst_sub_parse_init),
28830 (gst_sub_parse_set_property), (gst_sub_parse_get_property),
28831 (convert_encoding):
28832 * gst/subparse/gstsubparse.h:
28833 Add 'encoding' property (#341681).
28834 * gst/subparse/samiparse.c: (characters_sami):
28835 Output is pango markup, so we need to escape text
28836 between tags (#342143).
28838 2006-05-16 17:34:14 +0000 Tim-Philipp Müller <tim@centricular.net>
28840 gst-libs/gst/audio/multichannel.c: It's okay to have caps with channels=1 and a channel position different from GST_A...
28841 Original commit message from CVS:
28842 * gst-libs/gst/audio/multichannel.c:
28843 (gst_audio_check_channel_positions):
28844 It's okay to have caps with channels=1 and a channel position
28845 different from GST_AUDIO_CHANNEL_POSITION_FRONT_MONO
28846 (deinterleavers might want to keep the position in the caps,
28847 so that they can be re-interleaved again properly later).
28848 Leave check for unexpected 2-channel layouts intact for now.
28850 2006-05-16 16:28:10 +0000 Zaheer Abbas Merali <zaheerabbas@merali.org>
28852 gst/tcp/gsttcp.c: Return GST_FLOW_UNEXPECTED when we have an eos on the socket so basesrc can do its job correctly.
28853 Original commit message from CVS:
28854 2006-05-16 Zaheer Abbas Merali <zaheerabbas at merali dot org>
28855 * gst/tcp/gsttcp.c: (gst_tcp_socket_read):
28856 Return GST_FLOW_UNEXPECTED when we have an eos on the socket so
28857 basesrc can do its job correctly.
28859 2006-05-16 15:52:17 +0000 Tim-Philipp Müller <tim@centricular.net>
28861 ext/alsa/: Refactor and improve caps probing code: probe signedness when we probe the supported formats/widths; set e...
28862 Original commit message from CVS:
28863 * ext/alsa/Makefile.am:
28864 * ext/alsa/gstalsa.c: (gst_alsa_detect_rates),
28865 (gst_alsa_detect_formats), (get_channel_free_structure),
28866 (caps_add_channel_configuration), (gst_alsa_detect_channels),
28867 (gst_alsa_probe_supported_formats):
28868 * ext/alsa/gstalsa.h:
28869 * ext/alsa/gstalsasink.c: (gst_alsasink_getcaps):
28870 Refactor and improve caps probing code: probe signedness
28871 when we probe the supported formats/widths; set endianness
28872 to the one we actually probed for (ie. cpu endianness).
28873 * ext/alsa/gstalsasrc.c: (gst_alsasrc_init), (gst_alsasrc_getcaps),
28874 (gst_alsasrc_close):
28875 * ext/alsa/gstalsasrc.h:
28876 Implement caps probing for alsasrc.
28878 2006-05-15 17:42:19 +0000 Wim Taymans <wim.taymans@gmail.com>
28880 ext/theora/theoradec.c: Cleanups, add some G_LIKELY.
28881 Original commit message from CVS:
28882 * ext/theora/theoradec.c: (gst_theora_dec_reset),
28883 (theora_dec_src_query), (theora_dec_src_event),
28884 (theora_dec_sink_event), (theora_handle_comment_packet),
28885 (theora_handle_data_packet), (theora_dec_change_state):
28886 Cleanups, add some G_LIKELY.
28887 Use segment helpers instead of our own wrong code.
28888 Clear queued buffers on seek and READY.
28889 * ext/vorbis/vorbisdec.c: (gst_vorbis_dec_reset),
28890 (vorbis_dec_convert), (vorbis_dec_src_query),
28891 (vorbis_dec_src_event), (vorbis_dec_sink_event),
28892 (vorbis_handle_comment_packet), (vorbis_dec_push),
28893 (vorbis_handle_data_packet), (vorbis_dec_chain),
28894 (vorbis_dec_change_state):
28895 * ext/vorbis/vorbisdec.h:
28896 Remove old useless packetno variable.
28897 Do position query properly.
28899 Do cleanup of queued buffers in new helper function
28902 2006-05-15 17:17:22 +0000 Tim-Philipp Müller <tim@centricular.net>
28904 ext/alsa/gstalsasink.c: Query supported sample rates. Fixes #341732.
28905 Original commit message from CVS:
28906 * ext/alsa/gstalsasink.c: (gst_alsasink_getcaps):
28907 Query supported sample rates. Fixes #341732.
28909 2006-05-15 17:01:02 +0000 Julien Moutte <julien@moutte.net>
28911 gst/playback/gstdecodebin.c: Make decodebin reusable when going from PAUSE_TO_READY and then back to PAUSED.
28912 Original commit message from CVS:
28913 2006-05-15 Julien MOUTTE <julien@moutte.net>
28914 * gst/playback/gstdecodebin.c: (cleanup_decodebin),
28915 (gst_decode_bin_change_state): Make decodebin reusable
28916 when going from PAUSE_TO_READY and then back to PAUSED.
28919 2006-05-15 16:49:31 +0000 Wim Taymans <wim.taymans@gmail.com>
28921 ext/vorbis/vorbisdec.c: Cleanups. Use refcounting and DEBUG_OBJECT.
28922 Original commit message from CVS:
28923 * ext/vorbis/vorbisdec.c: (vorbis_get_query_types),
28924 (vorbis_dec_convert), (vorbis_dec_src_query),
28925 (vorbis_dec_sink_query), (vorbis_dec_src_event),
28926 (vorbis_dec_sink_event), (vorbis_handle_identification_packet),
28927 (vorbis_dec_clean_queued), (vorbis_dec_push),
28928 (vorbis_handle_data_packet), (vorbis_dec_change_state):
28929 Cleanups. Use refcounting and DEBUG_OBJECT.
28930 Reset segment on flush, use code methods instead of our
28932 Fix potential memleak.
28934 2006-05-15 16:46:44 +0000 Tim-Philipp Müller <tim@centricular.net>
28936 ext/alsa/gstalsasink.*: Don't leak allocated snd_output_t structure if there's more than one alsasink instance at a t...
28937 Original commit message from CVS:
28938 * ext/alsa/gstalsasink.c: (gst_alsasink_finalise),
28939 (gst_alsasink_init):
28940 * ext/alsa/gstalsasink.h:
28941 Don't leak allocated snd_output_t structure if there's
28942 more than one alsasink instance at a time (#341873).
28943 Also fix GObject macros in header file.
28945 2006-05-15 15:31:30 +0000 Tim-Philipp Müller <tim@centricular.net>
28947 gst/subparse/gstsubparse.c: Don't use libxml functions in the typefinding code.
28948 Original commit message from CVS:
28949 * gst/subparse/gstsubparse.c:
28950 (gst_sub_parse_data_format_autodetect):
28951 Don't use libxml functions in the typefinding code.
28953 2006-05-15 15:01:08 +0000 Wim Taymans <wim.taymans@gmail.com>
28955 ext/ogg/gstoggdemux.c: Fix seeking performance in the case where a non-header packet has a 0 granulepos (busted theor...
28956 Original commit message from CVS:
28957 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_packet):
28958 Fix seeking performance in the case where a non-header
28959 packet has a 0 granulepos (busted theora case).
28962 2006-05-15 14:19:35 +0000 Tim-Philipp Müller <tim@centricular.net>
28964 gst/subparse/gstsubparse.c: Improve SAMI typefinding: handle case where there are whitespaces or newlines in front of...
28965 Original commit message from CVS:
28966 * gst/subparse/gstsubparse.c:
28967 (gst_sub_parse_data_format_autodetect):
28968 Improve SAMI typefinding: handle case where there are
28969 whitespaces or newlines in front of the first <SAMI>
28972 2006-05-15 12:18:13 +0000 Tim-Philipp Müller <tim@centricular.net>
28974 configure.ac: Build video4linux plugin even if there's no XVIDEO, just without implementing the GstXOverlay interface...
28975 Original commit message from CVS:
28977 Build video4linux plugin even if there's no XVIDEO, just
28978 without implementing the GstXOverlay interface (#334002).
28980 2006-05-15 10:17:04 +0000 Tim-Philipp Müller <tim@centricular.net>
28982 Add tentative support for libvisual-0.4 (#336881).
28983 Original commit message from CVS:
28985 * ext/libvisual/visual.c: (gst_visual_actor_plugin_is_gl),
28987 Add tentative support for libvisual-0.4 (#336881).
28989 2006-05-15 09:41:03 +0000 Young-Ho Cha <ganadist@chollian.net>
28991 gst/subparse/samiparse.c: Need to map "silver" colour explicitly (#169936).
28992 Original commit message from CVS:
28993 Patch by: Young-Ho Cha <ganadist at chollian net>
28994 * gst/subparse/samiparse.c: (handle_start_font):
28995 Need to map "silver" colour explicitly (#169936).
28997 2006-05-15 09:14:35 +0000 Young-Ho Cha <ganadist@chollian.net>
28999 gst/subparse/: Add support for SAMI subtitles (#169936).
29000 Original commit message from CVS:
29001 Patch by: Young-Ho Cha <ganadist at chollian net>
29002 * gst/subparse/Makefile.am:
29003 * gst/subparse/gstsubparse.c: (gst_sub_parse_dispose),
29004 (parser_state_dispose), (gst_sub_parse_data_format_autodetect),
29005 (gst_sub_parse_format_autodetect), (feed_textbuf),
29006 (gst_subparse_type_find), (plugin_init):
29007 * gst/subparse/gstsubparse.h:
29008 * gst/subparse/samiparse.c:
29009 * gst/subparse/samiparse.h:
29010 Add support for SAMI subtitles (#169936).
29012 2006-05-14 21:18:58 +0000 Thomas Vander Stichele <thomas@apestaart.org>
29014 * win32/common/config.h:
29016 Original commit message from CVS:
29019 2006-05-14 21:18:16 +0000 Thomas Vander Stichele <thomas@apestaart.org>
29022 fix mistakes in README
29023 Original commit message from CVS:
29024 fix mistakes in README
29026 2006-05-14 18:15:17 +0000 Michael Smith <msmith@xiph.org>
29028 gst/audioconvert/gstchannelmix.c: Fix #341696: crash when mixing L+R+C to mono or stereo.
29029 Original commit message from CVS:
29030 * gst/audioconvert/gstchannelmix.c: (gst_channel_mix_fill_others):
29031 Fix #341696: crash when mixing L+R+C to mono or stereo.
29032 * tests/check/Makefile.am:
29033 * tests/check/elements/audioconvert.c: (set_channel_positions),
29034 (get_float_mc_caps), (get_int_mc_caps), (GST_START_TEST),
29035 (audioconvert_suite):
29036 Add test for the above, including some generic framework bits for
29037 testing multichannel things.
29039 2006-05-14 16:05:47 +0000 Jan Schmidt <thaytan@mad.scientist.com>
29043 Original commit message from CVS:
29046 === release 0.10.7 ===
29048 2006-05-14 16:00:57 +0000 Jan Schmidt <thaytan@mad.scientist.com>
29050 configure.ac: releasing 0.10.7, "Leave the gun"
29051 Original commit message from CVS:
29052 2006-05-14 Jan Schmidt <thaytan@mad.scientist.com>
29054 releasing 0.10.7, "Leave the gun"
29056 2006-05-14 15:55:16 +0000 Jan Schmidt <thaytan@mad.scientist.com>
29074 Original commit message from CVS:
29077 2006-05-12 22:22:37 +0000 Jan Schmidt <thaytan@mad.scientist.com>
29080 Original commit message from CVS:
29081 * gst/videoscale/gstvideoscale.c: (gst_video_scale_fixate_caps):
29082 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_setcaps):
29085 2006-05-12 21:30:00 +0000 Jan Schmidt <thaytan@mad.scientist.com>
29087 Fix integer overflow problem with pixel-aspect-ratio calculations in videoscale and xvimagesink (#341542)
29088 Original commit message from CVS:
29089 * docs/libs/gst-plugins-base-libs-docs.sgml:
29090 * docs/libs/gst-plugins-base-libs-sections.txt:
29091 * gst-libs/gst/video/video.c: (gst_video_calculate_display_ratio):
29092 * gst-libs/gst/video/video.h:
29093 * gst/videoscale/Makefile.am:
29094 * gst/videoscale/gstvideoscale.c: (gst_video_scale_fixate_caps):
29095 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_setcaps):
29096 * tests/check/Makefile.am:
29097 * tests/check/libs/video.c: (GST_START_TEST), (video_suite),
29099 Fix integer overflow problem with pixel-aspect-ratio calculations
29100 in videoscale and xvimagesink (#341542)
29102 2006-05-12 16:56:52 +0000 Tim-Philipp Müller <tim@centricular.net>
29104 gst-libs/gst/tag/gstid3tag.c: Map GST_IMAGE_TAG to and from ID2v2 APIC frames (#341557).
29105 Original commit message from CVS:
29106 * gst-libs/gst/tag/gstid3tag.c:
29107 Map GST_IMAGE_TAG to and from ID2v2 APIC frames (#341557).
29109 2006-05-12 10:39:08 +0000 Sébastien Moutte <sebastien@moutte.net>
29111 win32/MANIFEST: update win32 files listing
29112 Original commit message from CVS:
29114 update win32 files listing
29116 2006-05-11 21:47:01 +0000 Thomas Vander Stichele <thomas@apestaart.org>
29118 * tests/check/elements/multifdsink.c:
29119 disable failing check on gentoo64
29120 Original commit message from CVS:
29121 disable failing check on gentoo64
29123 2006-05-11 21:35:44 +0000 Thomas Vander Stichele <thomas@apestaart.org>
29125 * tests/check/elements/multifdsink.c:
29126 disable failing check on gentoo64
29127 Original commit message from CVS:
29128 disable failing check on gentoo64
29130 2006-05-11 21:20:20 +0000 Thomas Vander Stichele <thomas@apestaart.org>
29132 * tests/check/elements/multifdsink.c:
29133 macros show the correct line
29134 Original commit message from CVS:
29135 macros show the correct line
29137 2006-05-11 21:04:08 +0000 Thomas Vander Stichele <thomas@apestaart.org>
29139 * tests/check/elements/multifdsink.c:
29140 macros show the correct line
29141 Original commit message from CVS:
29142 macros show the correct line
29144 2006-05-11 21:01:05 +0000 Sjoerd Simons <sjoerd@luon.net>
29146 gst/playback/gstplaybasebin.*: API: GstPlayBaseBin::stream-info-value-array property use a more bindings-friendly way...
29147 Original commit message from CVS:
29148 2006-05-11 Thomas Vander Stichele <thomas at apestaart dot org>
29149 patch by: Sjoerd Simons (sjoerd@luon.net)
29150 * gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init),
29151 (group_create), (group_destroy), (add_stream),
29152 (gst_play_base_bin_get_property),
29153 (gst_play_base_bin_get_streaminfo_value_array):
29154 * gst/playback/gstplaybasebin.h:
29155 API: GstPlayBaseBin::stream-info-value-array property
29156 use a more bindings-friendly way of exposing streaminfo
29157 using a GValueArray. Tested in ipython.
29160 2006-05-11 19:44:06 +0000 Thomas Vander Stichele <thomas@apestaart.org>
29162 * tests/check/elements/multifdsink.c:
29163 fix some type warnings
29164 Original commit message from CVS:
29165 fix some type warnings
29167 2006-05-11 19:38:22 +0000 Wim Taymans <wim.taymans@gmail.com>
29169 gst/playback/gstdecodebin.c: Also catch queue underruns but don't do anything yet.
29170 Original commit message from CVS:
29171 * gst/playback/gstdecodebin.c: (try_to_link_1), (queue_enlarge),
29172 (queue_underrun_cb), (queue_filled_cb):
29173 Also catch queue underruns but don't do anything yet.
29174 Refactor and comment queue enlarging code a bit.
29175 * gst/playback/gstplaybasebin.c: (queue_overrun),
29176 (queue_threshold_reached), (queue_out_of_data),
29177 (gen_preroll_element):
29178 If a queue over/underruns check that we don't create nasty
29179 deadlocks when the min-threshold is not reached but the
29180 max-bytes is. In those cases disable max-bytes when we
29181 know that the queue is fed timed data.
29184 2006-05-11 18:06:18 +0000 Tim-Philipp Müller <tim@centricular.net>
29186 gst/playback/gstplaybin.c: Make playbin automatically plug an 'audioresample' element before the audio sink as well. ...
29187 Original commit message from CVS:
29188 * gst/playback/gstplaybin.c: (gen_audio_element):
29189 Make playbin automatically plug an 'audioresample'
29190 element before the audio sink as well. This solves
29191 problems with sinks that only accept a very specific
29192 sample rate, like esdsink (e.g. #340379).
29194 2006-05-11 16:04:28 +0000 Tim-Philipp Müller <tim@centricular.net>
29196 gst/playback/gstplaybasebin.c: Make http sources send special headers so that we receive icecast metadata if the http...
29197 Original commit message from CVS:
29198 * gst/playback/gstplaybasebin.c: (gen_source_element):
29199 Make http sources send special headers so that we receive
29200 icecast metadata if the http stream is an icecast stream
29201 (otherwise the server will just ignore them). This also
29202 means that from now on users will need the 'icydemux'
29203 element from gst-plugins-good installed if they want to
29204 listen to icecast radio streams. (#341432, #333657).
29206 2006-05-11 12:34:53 +0000 Thomas Vander Stichele <thomas@apestaart.org>
29208 * gst/tcp/gstmultifdsink.c:
29210 Original commit message from CVS:
29213 2006-05-11 11:40:39 +0000 Thomas Vander Stichele <thomas@apestaart.org>
29215 gst/tcp/gstmultifdsink.c: remove stupid example from docs - it should come with a simple
29216 Original commit message from CVS:
29217 * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_remove_client_link),
29218 (gst_multi_fd_sink_new_client), (gst_multi_fd_sink_stop):
29219 remove stupid example from docs - it should come with a simple
29222 * tests/check/elements/multifdsink.c: (wait_bytes_served),
29223 (fail_if_can_read), (GST_START_TEST),
29224 (gst_multifdsink_create_streamheader), (multifdsink_suite):
29225 add a test for changing streamheader which exposes a bug in
29228 2006-05-11 10:33:46 +0000 Michael Smith <msmith@xiph.org>
29230 ext/gnomevfs/gstgnomevfssrc.*: Don't set icy-caps unless we have a sane interval value. Move interval to a local vari...
29231 Original commit message from CVS:
29232 * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_init),
29233 (gst_gnome_vfs_src_received_headers_callback):
29234 * ext/gnomevfs/gstgnomevfssrc.h:
29235 Don't set icy-caps unless we have a sane interval value. Move
29236 interval to a local variable; we never use it outside this function.
29238 2006-05-11 10:14:20 +0000 Wim Taymans <wim.taymans@gmail.com>
29240 sys/: Register special buffer types along with the objects so that they are not registered at runtime from N differen...
29241 Original commit message from CVS:
29242 * sys/ximage/ximagesink.c: (gst_ximagesink_get_type):
29243 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_type):
29244 Register special buffer types along with the objects so
29245 that they are not registered at runtime from N different
29246 streaming threads since they are not threadsafe.
29248 2006-05-10 18:31:27 +0000 Thomas Vander Stichele <thomas@apestaart.org>
29250 * tests/check/elements/multifdsink.c:
29251 set caps and plug leaks
29252 Original commit message from CVS:
29253 set caps and plug leaks
29255 2006-05-10 18:16:23 +0000 Thomas Vander Stichele <thomas@apestaart.org>
29257 tests/check/elements/multifdsink.c: add two more tests, one doing streamheader
29258 Original commit message from CVS:
29259 * tests/check/elements/multifdsink.c: (wait_bytes_served),
29260 (GST_START_TEST), (fail_unless_read), (multifdsink_suite):
29261 add two more tests, one doing streamheader
29263 2006-05-10 16:34:06 +0000 Thomas Vander Stichele <thomas@apestaart.org>
29265 gst/tcp/gstmultifdsink.c: clean up the bufqueue when shutting down
29266 Original commit message from CVS:
29267 * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_stop):
29268 clean up the bufqueue when shutting down
29269 * tests/check/Makefile.am:
29270 * tests/check/elements/multifdsink.c: (setup_multifdsink),
29271 (cleanup_multifdsink), (GST_START_TEST), (multifdsink_suite),
29273 add a test for the leak that was just fixed
29275 2006-05-10 15:16:33 +0000 Thomas Vander Stichele <thomas@apestaart.org>
29277 * gst/tcp/gstmultifdsink.c:
29279 Original commit message from CVS:
29282 2006-05-10 15:14:30 +0000 Thomas Vander Stichele <thomas@apestaart.org>
29284 * gst/tcp/gstmultifdsink.c:
29285 * gst/tcp/gstmultifdsink.h:
29287 Original commit message from CVS:
29290 2006-05-10 11:54:36 +0000 Wim Taymans <wim.taymans@gmail.com>
29292 gst/adder/gstadder.*: Updated some docs. Added comments and FIXMEs all over the place.
29293 Original commit message from CVS:
29294 * gst/adder/gstadder.c: (gst_adder_setcaps),
29295 (gst_adder_query_duration), (gst_adder_query), (forward_event),
29296 (gst_adder_src_event), (gst_adder_sink_event),
29297 (gst_adder_class_init), (gst_adder_finalize),
29298 (gst_adder_request_new_pad), (gst_adder_collected):
29299 * gst/adder/gstadder.h:
29300 Updated some docs. Added comments and FIXMEs all over the place.
29301 Improve debugging info.
29302 Fix leak on finalize by not calling the parent.
29303 Implement duration query.
29304 Make event forwarding threadsafe.
29305 Correctly send NEWSEGMENT at start and after flush.
29306 Handle EOS correctly.
29307 Post error when not negotiated.
29308 * tests/check/elements/adder.c: (GST_START_TEST):
29309 Added FIXME in the test.
29311 2006-05-09 19:24:46 +0000 Tim-Philipp Müller <tim@centricular.net>
29313 Const-ify GEnumValue and GFlagsValue arrays. Use
29314 Original commit message from CVS:
29315 * ext/pango/gsttextoverlay.c: (gst_text_overlay_valign_get_type),
29316 (gst_text_overlay_halign_get_type),
29317 (gst_text_overlay_wrap_mode_get_type):
29318 * ext/theora/theoradec.c: (theora_handle_type_packet),
29319 (theora_handle_data_packet):
29320 * ext/theora/theoraenc.c: (gst_border_mode_get_type),
29321 (theora_enc_sink_setcaps), (theora_enc_chain):
29322 * gst-libs/gst/cdda/gstcddabasesrc.c:
29323 (gst_cdda_base_src_mode_get_type):
29324 * gst/audiotestsrc/gstaudiotestsrc.c:
29325 (gst_audiostestsrc_wave_get_type):
29326 * gst/playback/gststreaminfo.c: (gst_stream_type_get_type):
29327 * gst/tcp/gstfdset.c: (gst_fdset_mode_get_type):
29328 * gst/tcp/gstmultifdsink.c: (gst_recover_policy_get_type),
29329 (gst_sync_method_get_type), (gst_unit_type_get_type),
29330 (gst_client_status_get_type):
29331 * gst/videoscale/gstvideoscale.c:
29332 (gst_video_scale_method_get_type):
29333 * gst/videotestsrc/gstvideotestsrc.c:
29334 (gst_video_test_src_pattern_get_type):
29335 * gst/videotestsrc/videotestsrc.c: (paint_setup_I420),
29336 (paint_setup_YV12), (paint_setup_YUY2), (paint_setup_UYVY),
29337 (paint_setup_YVYU), (paint_setup_IYU2), (paint_setup_Y41B),
29338 (paint_setup_Y42B), (paint_setup_Y800), (paint_setup_YVU9),
29339 (paint_setup_YUV9), (paint_setup_RGB888), (paint_setup_BGR888),
29340 (paint_setup_RGB565), (paint_setup_xRGB1555):
29341 Const-ify GEnumValue and GFlagsValue arrays. Use
29342 GST_ROUND_UP_* macros instead of home-made ones.
29344 2006-05-09 17:40:41 +0000 Tim-Philipp Müller <tim@centricular.net>
29346 configure.ac: Require core CVS for the new newsegment stuff.
29347 Original commit message from CVS:
29349 Require core CVS for the new newsegment stuff.
29351 2006-05-09 17:30:48 +0000 Sjoerd Simons <sjoerd@luon.net>
29353 gst/tcp/gstmultifdsink.c: Register nick for enum value (#341160).
29354 Original commit message from CVS:
29355 Patch by: Sjoerd Simons <sjoerd at luon net>
29356 * gst/tcp/gstmultifdsink.c: (gst_client_status_get_type):
29357 Register nick for enum value (#341160).
29359 2006-05-09 16:46:23 +0000 Stefan Kost <ensonic@users.sourceforge.net>
29361 gst/typefind/gsttypefindfunctions.c: backout typefind patch #340375
29362 Original commit message from CVS:
29363 * gst/typefind/gsttypefindfunctions.c: (m4a_type_find),
29365 backout typefind patch #340375
29366 * tests/check/elements/adder.c: (message_received),
29367 (GST_START_TEST), (adder_suite):
29368 redo, signal-handling of test
29370 2006-05-09 16:14:49 +0000 Wim Taymans <wim.taymans@gmail.com>
29372 gst/adder/gstadder.*: Remove bogus segment merging and forwarding, we don't care about timestamps anyway and we just ...
29373 Original commit message from CVS:
29374 * gst/adder/gstadder.c: (gst_adder_request_new_pad),
29375 (gst_adder_collected):
29376 * gst/adder/gstadder.h:
29377 Remove bogus segment merging and forwarding, we don't
29378 care about timestamps anyway and we just produce a
29380 Also create a nice NEWSEGMENT event when we start.
29381 Use _scale_int some more.
29383 2006-05-09 11:59:13 +0000 Edward Hervey <bilboed@bilboed.com>
29385 tests/icles/stress-xoverlay.c: Fix if core was built without parsing support.
29386 Original commit message from CVS:
29387 * tests/icles/stress-xoverlay.c:
29388 Fix if core was built without parsing support.
29390 2006-05-09 11:37:22 +0000 Tim-Philipp Müller <tim@centricular.net>
29392 gst-libs/gst/riff/riff-media.c: Add SEDG (Samsung MPEG-4) fourcc.
29393 Original commit message from CVS:
29394 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
29395 Add SEDG (Samsung MPEG-4) fourcc.
29397 2006-05-09 11:31:47 +0000 Edward Hervey <bilboed@bilboed.com>
29399 tests/examples/volume/volume.c: Fox if core was built without parsing support.
29400 Original commit message from CVS:
29401 * tests/examples/volume/volume.c:
29402 Fox if core was built without parsing support.
29403 * tests/examples/seek/seek.c:
29404 Disable the parse_launch example if core was built without parsing
29407 2006-05-09 11:21:24 +0000 Edward Hervey <bilboed@bilboed.com>
29409 tests/examples/seek/seek.c: Disable the parse_launch example if core was built without parsing support.
29410 Original commit message from CVS:
29411 * tests/examples/seek/seek.c:
29412 Disable the parse_launch example if core was built without parsing
29415 2006-05-08 15:51:15 +0000 Thomas Vander Stichele <thomas@apestaart.org>
29417 * docs/libs/tmpl/gstcolorbalance.sgml:
29418 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
29419 * gst/tcp/gstmultifdsink.c:
29420 * gst/videoscale/gstvideoscale.c:
29421 doc reparagraphing and DEBUG_FUNCPTRing
29422 Original commit message from CVS:
29423 doc reparagraphing and DEBUG_FUNCPTRing
29425 2006-05-08 11:51:43 +0000 Edward Hervey <bilboed@bilboed.com>
29427 autogen.sh: libtoolize on Darwin/MacOSX is called glibtoolize
29428 Original commit message from CVS:
29429 * autogen.sh: (CONFIGURE_DEF_OPT):
29430 libtoolize on Darwin/MacOSX is called glibtoolize
29432 2006-05-07 17:39:04 +0000 Stefan Kost <ensonic@users.sourceforge.net>
29434 tests/check/: Disable the adder test, until the build-slaves posses the kindness to either like it or to give valid r...
29435 Original commit message from CVS:
29436 * tests/check/Makefile.am:
29437 * tests/check/elements/adder.c: (event_loop), (GST_START_TEST):
29438 Disable the adder test, until the build-slaves posses the kindness to
29439 either like it or to give valid reason for not doing so
29441 2006-05-07 17:25:56 +0000 Stefan Kost <ensonic@users.sourceforge.net>
29443 tests/check/elements/adder.c: Shuffle NULL state change around and raise timeout more
29444 Original commit message from CVS:
29445 * tests/check/elements/adder.c: (event_loop), (GST_START_TEST),
29447 Shuffle NULL state change around and raise timeout more
29449 2006-05-07 17:07:03 +0000 Stefan Kost <ensonic@users.sourceforge.net>
29451 gst/typefind/gsttypefindfunctions.c: Add typefind to distinguish between "audio/x-m4a" and new type "video/mp4". Fixe...
29452 Original commit message from CVS:
29453 * gst/typefind/gsttypefindfunctions.c: (mp4_find_box),
29454 (mp4_type_find), (plugin_init):
29455 Add typefind to distinguish between "audio/x-m4a" and new type
29456 "video/mp4". Fixes #340375
29457 * tests/check/elements/adder.c: (adder_suite):
29458 Raise timeout to make buildbot happy
29460 2006-05-07 16:39:36 +0000 Stefan Kost <ensonic@users.sourceforge.net>
29462 Add sink-event handling to adder. It tries to merge incomming newsegment-events. Added test to check if segment_done ...
29463 Original commit message from CVS:
29464 * gst/adder/gstadder.c: (gst_adder_sink_event),
29465 (gst_adder_request_new_pad), (gst_adder_change_state):
29466 * gst/adder/gstadder.h:
29467 * tests/check/Makefile.am:
29468 * tests/check/elements/adder.c: (event_loop), (GST_START_TEST),
29469 (adder_suite), (main):
29470 Add sink-event handling to adder. It tries to merge incomming
29471 newsegment-events. Added test to check if segment_done is comming
29474 2006-05-05 16:34:15 +0000 Andy Wingo <wingo@pobox.com>
29477 * ext/theora/theoraparse.c:
29478 * ext/vorbis/vorbisparse.c:
29479 ext/theora/theoraparse.c (gst_theora_parse_init) ext/vorbis/vorbisparse.c (gst_vorbis_parse_init)
29480 Original commit message from CVS:
29481 2006-05-05 Andy Wingo <wingo@pobox.com>
29482 * ext/theora/theoraparse.c (gst_theora_parse_init)
29483 (theora_parse_src_convert, theora_parse_src_query):
29484 * ext/vorbis/vorbisparse.c (gst_vorbis_parse_init)
29485 (vorbis_parse_convert, vorbis_parse_src_query): Add convert and
29486 query functions on the source pads of the theora and vorbis parse
29487 elements. Fixes position querying when doing a remux.
29489 2006-05-05 13:46:37 +0000 Michael Smith <msmith@xiph.org>
29491 ext/theora/theoraparse.c: Fix flushing.
29492 Original commit message from CVS:
29493 * ext/theora/theoraparse.c: (parse_granulepos),
29494 (theora_parse_drain_queue_prematurely),
29495 (theora_parse_queue_buffer), (theora_parse_sink_event):
29497 Fix invalid granulepos outputs when starting with a non-keyframe.
29499 2006-05-05 12:37:35 +0000 Jan Schmidt <thaytan@mad.scientist.com>
29501 gst/typefind/gsttypefindfunctions.c: Rearrange MPEG system stream detection, fixing some memleaks in the process.
29502 Original commit message from CVS:
29503 * gst/typefind/gsttypefindfunctions.c: (mpeg2_sys_type_find),
29504 (mpeg1_sys_type_find), (ogganx_type_find), (sw_data_destroy):
29505 Rearrange MPEG system stream detection, fixing some memleaks in the
29507 Constify the data for STARTS_WITH and RIFF helper handlers. Make sure
29508 they clean up their data correctly.
29509 Remove unused ogganx caps and move the 'is_annodex' check to inside
29510 the 'is_ogg' if statement.
29512 2006-05-05 11:33:37 +0000 Wim Taymans <wim.taymans@gmail.com>
29514 gst/playback/gstdecodebin.c: Properly remove ghostpads. Fixes #340392
29515 Original commit message from CVS:
29516 * gst/playback/gstdecodebin.c: (cleanup_decodebin):
29517 Properly remove ghostpads. Fixes #340392
29519 2006-05-04 18:43:58 +0000 David Schleef <ds@schleef.org>
29521 gst/typefind/gsttypefindfunctions.c:
29522 Original commit message from CVS:
29523 * gst/typefind/gsttypefindfunctions.c:
29525 2006-05-03 16:32:19 +0000 Jan Schmidt <thaytan@mad.scientist.com>
29527 gst/typefind/gsttypefindfunctions.c: When typefinding an MP3 in push-based mode, don't penalise the probability down ...
29528 Original commit message from CVS:
29529 * gst/typefind/gsttypefindfunctions.c: (mp3_type_find_at_offset),
29530 (mpeg_ts_probe_headers), (mpeg_ts_type_find):
29531 When typefinding an MP3 in push-based mode, don't penalise the
29532 probability down to 74% when we found 5 valid frames just because we
29533 can't peek the end of the file.
29534 Make the probability for detecting MPEG Transport Streams based on the
29535 number of sequential headers we successfully detected.
29537 2006-05-03 15:52:46 +0000 Wim Taymans <wim.taymans@gmail.com>
29539 ext/vorbis/vorbisdec.c: Still produce an error when we receive an empty packet.
29540 Original commit message from CVS:
29541 * ext/vorbis/vorbisdec.c: (vorbis_dec_sink_event),
29542 (vorbis_dec_push), (vorbis_dec_chain):
29543 Still produce an error when we receive an empty packet.
29545 2006-05-03 15:34:48 +0000 Wim Taymans <wim.taymans@gmail.com>
29547 ext/ogg/gstoggdemux.c: Mark buffers with DISCONT after seek and after activating new chains.
29548 Original commit message from CVS:
29549 * ext/ogg/gstoggdemux.c: (gst_ogg_demux_chain_peer),
29550 (gst_ogg_chain_mark_discont), (gst_ogg_chain_new_stream),
29551 (gst_ogg_demux_activate_chain), (gst_ogg_demux_perform_seek):
29552 Mark buffers with DISCONT after seek and after activating new
29554 * ext/theora/gsttheoradec.h:
29555 * ext/theora/theoradec.c: (gst_theora_dec_reset),
29556 (theora_get_query_types), (theora_dec_sink_event),
29557 (theora_dec_push), (theora_handle_data_packet), (theora_dec_chain),
29558 (theora_dec_change_state):
29560 Detect and mark DISCONT buffers.
29561 * ext/vorbis/vorbisdec.c: (vorbis_dec_src_query),
29562 (vorbis_dec_sink_event), (vorbis_dec_push), (vorbis_dec_chain),
29563 (vorbis_dec_change_state):
29564 * ext/vorbis/vorbisdec.h:
29566 Detect and mark DISCONT buffers.
29567 Don't crash on 0 sized buffers.
29569 2006-05-03 08:58:13 +0000 Wim Taymans <wim.taymans@gmail.com>
29571 gst/volume/gstvolume.c: Increase "volume" property to 10.0. Fixes #340369.
29572 Original commit message from CVS:
29573 * gst/volume/gstvolume.c: (volume_funcfind), (volume_set_caps),
29574 (volume_transform_ip):
29575 Increase "volume" property to 10.0. Fixes #340369.
29576 Set the process function to NULL when capsnego fails so that
29577 we properly error out.
29579 2006-05-02 18:15:25 +0000 Stefan Kost <ensonic@users.sourceforge.net>
29581 gst/playback/: free cpas using gst_caps_unref, don't leak caps-strings
29582 Original commit message from CVS:
29583 * gst/playback/gstplaybin.c: (add_sink):
29584 * gst/playback/test.c: (main):
29585 * gst/playback/test5.c: (dump_element_stats):
29586 * gst/playback/test6.c: (main):
29587 free cpas using gst_caps_unref, don't leak caps-strings
29589 2006-05-02 06:33:54 +0000 Thomas Vander Stichele <thomas@apestaart.org>
29591 * gst-libs/gst/rtp/gstbasertppayload.c:
29593 Original commit message from CVS:
29596 2006-05-01 19:08:40 +0000 Tim-Philipp Müller <tim@centricular.net>
29598 gst/typefind/gsttypefindfunctions.c: Refine musepack typefinding a bit. Return MAXIMUM probability when we detect str...
29599 Original commit message from CVS:
29600 * gst/typefind/gsttypefindfunctions.c: (musepack_type_find),
29602 Refine musepack typefinding a bit. Return MAXIMUM
29603 probability when we detect stream version 7 to make
29604 sure the mpeg audio typefinder doesn't trump us.
29606 2006-04-29 16:25:58 +0000 Tim-Philipp Müller <tim@centricular.net>
29608 gst-libs/gst/riff/riff-media.c: Protect against unexpected NULL strf_data buffer.
29609 Original commit message from CVS:
29610 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
29611 Protect against unexpected NULL strf_data buffer.
29613 2006-04-29 13:09:52 +0000 Thomas Vander Stichele <thomas@apestaart.org>
29615 tests/check/elements/audioconvert.c: interpret the out[] buffer in the order the bytes are actually put in, which is ...
29616 Original commit message from CVS:
29617 * tests/check/elements/audioconvert.c: (verify_convert),
29619 interpret the out[] buffer in the order the bytes are actually
29620 put in, which is LITTLE_ENDIAN, not BYTE_ORDER.
29621 Other tests should use BYTE_ORDER since the array is filled in
29624 2006-04-29 12:10:52 +0000 Thomas Vander Stichele <thomas@apestaart.org>
29626 * tests/check/elements/audioconvert.c:
29627 dump expected data when audioconvert test fails
29628 Original commit message from CVS:
29629 dump expected data when audioconvert test fails
29631 2006-04-29 11:55:09 +0000 Thomas Vander Stichele <thomas@apestaart.org>
29633 tests/check/elements/audioconvert.c: when a test fails, give an indication of which it is
29634 Original commit message from CVS:
29635 * tests/check/elements/audioconvert.c: (verify_convert),
29637 when a test fails, give an indication of which it is
29639 2006-04-29 09:48:16 +0000 Thomas Vander Stichele <thomas@apestaart.org>
29641 * ext/ogg/gstoggmux.c:
29642 * ext/theora/theoraenc.c:
29643 add another include
29644 Original commit message from CVS:
29645 add another include
29647 2006-04-29 01:24:31 +0000 Thomas Vander Stichele <thomas@apestaart.org>
29649 * gst/subparse/gstssaparse.c:
29650 atoi() needs stdlib.h
29651 Original commit message from CVS:
29652 atoi() needs stdlib.h
29654 2006-04-29 01:18:05 +0000 Thomas Vander Stichele <thomas@apestaart.org>
29656 * gst/playback/test4.c:
29657 * gst/playback/test5.c:
29658 * gst/playback/test6.c:
29659 exit needs stdlib.h
29660 Original commit message from CVS:
29661 exit needs stdlib.h
29663 2006-04-29 01:10:34 +0000 Thomas Vander Stichele <thomas@apestaart.org>
29665 gst-libs/gst/cdda/gstcddabasesrc.c: compile fix; strtol() needs <stdlib.h>
29666 Original commit message from CVS:
29667 * gst-libs/gst/cdda/gstcddabasesrc.c:
29668 compile fix; strtol() needs <stdlib.h>
29670 2006-04-29 01:04:51 +0000 Thomas Vander Stichele <thomas@apestaart.org>
29674 * docs/Makefile.am:
29675 * docs/libs/Makefile.am:
29676 * docs/libs/tmpl/gstcolorbalance.sgml:
29677 * docs/plugins/Makefile.am:
29679 use common upload.mak
29680 Original commit message from CVS:
29681 use common upload.mak
29683 2006-04-28 19:46:37 +0000 Stefan Kost <ensonic@users.sourceforge.net>
29685 make GstElementDetails const
29686 Original commit message from CVS:
29687 * ext/alsa/gstalsamixerelement.c:
29688 * ext/alsa/gstalsasrc.c:
29689 * ext/cdparanoia/gstcdparanoiasrc.c:
29690 * ext/gnomevfs/gstgnomevfssink.c:
29691 * ext/gnomevfs/gstgnomevfssrc.c:
29692 * ext/ogg/gstoggdemux.c:
29693 * ext/ogg/gstoggmux.c:
29694 * ext/ogg/gstoggparse.c:
29695 * ext/ogg/gstogmparse.c:
29696 * ext/pango/gstclockoverlay.c:
29697 * ext/pango/gsttextoverlay.c:
29698 * ext/pango/gsttextrender.c:
29699 * ext/pango/gsttimeoverlay.c:
29700 * ext/theora/theoradec.c:
29701 * ext/theora/theoraenc.c:
29702 * ext/vorbis/vorbisdec.c:
29703 * ext/vorbis/vorbisenc.c:
29704 * gst-libs/gst/audio/gstaudiofilter.c:
29705 * gst-libs/gst/audio/gstaudiofiltertemplate.c:
29706 * gst/audioconvert/gstaudioconvert.c:
29707 * gst/audiorate/gstaudiorate.c:
29708 * gst/audioresample/gstaudioresample.c:
29709 * gst/audiotestsrc/gstaudiotestsrc.c:
29710 * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
29711 * gst/playback/gstdecodebin.c:
29712 * gst/playback/gstplaybin.c:
29713 * gst/playback/gststreamselector.c:
29714 * gst/subparse/gstsubparse.c:
29715 * gst/tcp/gstmultifdsink.c:
29716 * gst/tcp/gsttcpclientsink.c:
29717 * gst/tcp/gsttcpclientsrc.c:
29718 * gst/tcp/gsttcpserversink.c:
29719 * gst/tcp/gsttcpserversrc.c:
29720 * gst/typefind/gsttypefindfunctions.c: (plugin_init):
29721 * gst/videorate/gstvideorate.c:
29722 * gst/videoscale/gstvideoscale.c:
29723 * gst/videotestsrc/gstvideotestsrc.c:
29724 * gst/volume/gstvolume.c:
29725 * sys/v4l/gstv4ljpegsrc.c:
29726 * sys/v4l/gstv4lmjpegsink.c:
29727 * sys/v4l/gstv4lmjpegsrc.c:
29728 * sys/v4l/gstv4lsrc.c:
29729 * sys/ximage/ximagesink.c:
29730 * sys/xvimage/xvimagesink.c:
29731 * tests/check/libs/cddabasesrc.c:
29732 make GstElementDetails const
29734 2006-04-28 19:17:06 +0000 Stefan Kost <ensonic@users.sourceforge.net>
29736 gst/adder/gstadder.c: send events from src-pad to all sink-pads fixes #338657
29737 Original commit message from CVS:
29738 * gst/adder/gstadder.c: (gst_adder_setcaps), (gst_adder_src_event),
29740 send events from src-pad to all sink-pads fixes #338657
29742 2006-04-28 19:08:34 +0000 Stefan Kost <ensonic@users.sourceforge.net>
29744 ext/alsa/gstalsasink.c: query witdh capabilities from alsa, fixes #338919
29745 Original commit message from CVS:
29746 * ext/alsa/gstalsasink.c: (gst_alsasink_getcaps),
29747 (alsasink_parse_spec):
29748 query witdh capabilities from alsa, fixes #338919
29750 2006-04-28 15:31:28 +0000 Wim Taymans <wim.taymans@gmail.com>
29752 gst/tcp/gstmultifdsink.*: Fix race condition in multifdsink that can lead to spurious duplicate clients. this patch a...
29753 Original commit message from CVS:
29754 * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init),
29755 (gst_multi_fd_sink_remove_client_link):
29756 * gst/tcp/gstmultifdsink.h:
29757 Fix race condition in multifdsink that can lead to spurious
29758 duplicate clients. this patch adds a new signal that is fired when
29759 multifdsink has removed all references to the fd.
29761 Updated documentation.
29762 API: client-fd-removed signal added
29764 2006-04-28 15:24:00 +0000 Michael Smith <msmith@xiph.org>
29766 gst/tcp/gstmultifdsink.c: When asking g_value_array_new to prealloc elements, we may as well ask for the right number...
29767 Original commit message from CVS:
29768 * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_get_stats):
29769 When asking g_value_array_new to prealloc elements, we may as well
29770 ask for the right number of elements.
29772 2006-04-28 15:08:09 +0000 Wim Taymans <wim.taymans@gmail.com>
29774 gst-libs/gst/audio/gstbaseaudiosink.c: patch to make timestamp checking more tollerant to rounding errors given that ...
29775 Original commit message from CVS:
29776 * gst-libs/gst/audio/gstbaseaudiosink.c:
29777 (gst_base_audio_sink_drain), (gst_base_audio_sink_event),
29778 (gst_base_audio_sink_render), (gst_base_audio_sink_change_state):
29779 patch to make timestamp checking more tollerant to rounding
29780 errors given that real discontinuities are to be marked on
29781 buffers. Fixes some asf files and #338778.
29782 Also avoid some crashers when we receive an event in the
29785 2006-04-28 15:01:58 +0000 Michael Smith <msmith@xiph.org>
29787 ext/gnomevfs/gstgnomevfssrc.*: Remove ICY handling (mostly) from gnomevfssrc, in favour of proper shared support with...
29788 Original commit message from CVS:
29789 * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_class_init),
29790 (gst_gnome_vfs_src_init), (gst_gnome_vfs_src_finalize),
29791 (gst_gnome_vfs_src_get_property),
29792 (gst_gnome_vfs_src_send_additional_headers_callback),
29793 (gst_gnome_vfs_src_received_headers_callback),
29794 (gst_gnome_vfs_src_create), (gst_gnome_vfs_src_start),
29795 (gst_gnome_vfs_src_stop):
29796 * ext/gnomevfs/gstgnomevfssrc.h:
29797 Remove ICY handling (mostly) from gnomevfssrc, in favour of
29798 proper shared support within icydemux.
29800 2006-04-28 14:49:22 +0000 Thomas Vander Stichele <thomas@apestaart.org>
29802 gst/videorate/gstvideorate.c: fix up docs fix a leak when no caps negotiated fix counting of input frames
29803 Original commit message from CVS:
29804 * gst/videorate/gstvideorate.c: (gst_video_rate_reset),
29805 (gst_video_rate_swap_prev), (gst_video_rate_chain):
29807 fix a leak when no caps negotiated
29808 fix counting of input frames
29809 * tests/check/elements/.cvsignore:
29810 * tests/check/elements/videorate.c: (assert_videorate_stats),
29811 (GST_START_TEST), (videorate_suite):
29812 add tests for these
29814 2006-04-28 14:48:11 +0000 Wim Taymans <wim.taymans@gmail.com>
29816 gst-libs/gst/audio/gstringbuffer.c: Check arguments passed to public functions instead of crashing.
29817 Original commit message from CVS:
29818 * gst-libs/gst/audio/gstringbuffer.c:
29819 (gst_ring_buffer_set_callback), (gst_ring_buffer_acquire),
29820 (gst_ring_buffer_release), (gst_ring_buffer_is_acquired),
29821 (gst_ring_buffer_set_flushing), (gst_ring_buffer_start),
29822 (gst_ring_buffer_pause), (gst_ring_buffer_stop),
29823 (gst_ring_buffer_delay), (gst_ring_buffer_samples_done),
29824 (gst_ring_buffer_set_sample), (gst_ring_buffer_clear_all),
29825 (gst_ring_buffer_commit), (gst_ring_buffer_read),
29826 (gst_ring_buffer_prepare_read), (gst_ring_buffer_advance),
29827 (gst_ring_buffer_clear), (gst_ring_buffer_may_start):
29828 Check arguments passed to public functions instead of
29831 2006-04-28 14:37:46 +0000 Wim Taymans <wim.taymans@gmail.com>
29833 gst-libs/gst/audio/gstbaseaudiosrc.c: GstBaseAudioSrc must be live or it does not work.
29834 Original commit message from CVS:
29835 * gst-libs/gst/audio/gstbaseaudiosrc.c: (gst_base_audio_src_init),
29836 (gst_base_audio_src_get_time), (gst_base_audio_src_create):
29837 GstBaseAudioSrc must be live or it does not work.
29838 * gst-libs/gst/audio/gstaudiosrc.c: (gst_audio_src_init):
29839 Don't set live to TRUE as this is the default in the parentclass.
29841 2006-04-28 14:37:12 +0000 Thomas Vander Stichele <thomas@apestaart.org>
29843 * win32/common/config.h:
29845 Original commit message from CVS:
29848 2006-04-28 14:33:45 +0000 Wim Taymans <wim.taymans@gmail.com>
29850 gst/videoscale/gstvideoscale.c: Videoscale doesn't pass on pixel-aspect ratio. Handle all fixation cases better. Fixe...
29851 Original commit message from CVS:
29852 * gst/videoscale/gstvideoscale.c: (gst_video_scale_transform_caps),
29853 (gst_video_scale_fixate_caps), (gst_video_scale_src_event):
29854 Videoscale doesn't pass on pixel-aspect ratio. Handle all
29855 fixation cases better. Fixes #338991
29857 2006-04-28 14:24:38 +0000 Wim Taymans <wim.taymans@gmail.com>
29859 gst/videotestsrc/gstvideotestsrc.c: Handle 0/1 framerate correctly Fixes #331901.
29860 Original commit message from CVS:
29861 * gst/videotestsrc/gstvideotestsrc.c: (gst_video_test_src_create):
29862 Handle 0/1 framerate correctly Fixes #331901.
29864 2006-04-28 14:22:16 +0000 Wim Taymans <wim.taymans@gmail.com>
29866 tests/check/elements/audioconvert.c: Added check for correct clipping when doing float samples in audioconvert.
29867 Original commit message from CVS:
29868 * tests/check/elements/audioconvert.c: (get_float_caps),
29869 (GST_START_TEST), (audioconvert_suite):
29870 Added check for correct clipping when doing float samples
29873 2006-04-28 14:19:49 +0000 Wim Taymans <wim.taymans@gmail.com>
29875 gst/videorate/gstvideorate.c: Print more debugging info.
29876 Original commit message from CVS:
29877 * gst/videorate/gstvideorate.c: (gst_video_rate_event),
29878 (gst_video_rate_chain):
29879 Print more debugging info.
29881 2006-04-28 14:17:00 +0000 Wim Taymans <wim.taymans@gmail.com>
29883 gst/audioresample/gstaudioresample.c: Add support for other formats audioresample can handle such as 32 bits in and f...
29884 Original commit message from CVS:
29885 * gst/audioresample/gstaudioresample.c: (gst_audioresample_init),
29886 (resample_set_state_from_caps):
29887 Add support for other formats audioresample can handle such as
29888 32 bits in and float and 64 bits float. Fixes #301759
29890 2006-04-28 14:12:28 +0000 Wim Taymans <wim.taymans@gmail.com>
29892 gst/audioconvert/audioconvert.c: correctly clip float samples > 1.0. Fixes #338718
29893 Original commit message from CVS:
29894 * gst/audioconvert/audioconvert.c: (float):
29895 correctly clip float samples > 1.0. Fixes #338718
29897 2006-04-28 13:35:34 +0000 Young-Ho Cha <ganadist@chollian.net>
29899 ext/pango/gsttextoverlay.c: Don't strip newlines from the text. Also, center lines within multi-line paragraphs (#339...
29900 Original commit message from CVS:
29901 Patch by: Young-Ho Cha <ganadist at chollian net>
29902 * ext/pango/gsttextoverlay.c: (gst_text_overlay_init),
29903 (gst_text_overlay_render_text):
29904 Don't strip newlines from the text. Also, center lines
29905 within multi-line paragraphs (#339405).
29907 2006-04-28 12:15:33 +0000 Tim-Philipp Müller <tim@centricular.net>
29909 gst/typefind/gsttypefindfunctions.c: Fix wavpack typefinding to work in more cases (don't peek for chunks of multiple...
29910 Original commit message from CVS:
29911 * gst/typefind/gsttypefindfunctions.c: (wavpack_type_find):
29912 Fix wavpack typefinding to work in more cases (don't peek
29913 for chunks of multiple hundred kBs at once, but process
29914 things step-by-step in smaller units). Fixes #339786.
29916 2006-04-28 10:58:41 +0000 Thomas Vander Stichele <thomas@apestaart.org>
29921 Original commit message from CVS:
29924 === release 0.10.6 ===
29926 2006-04-28 10:53:40 +0000 Thomas Vander Stichele <thomas@apestaart.org>
29932 * docs/plugins/gst-plugins-base-plugins.signals:
29933 * docs/plugins/inspect/plugin-adder.xml:
29934 * docs/plugins/inspect/plugin-alsa.xml:
29935 * docs/plugins/inspect/plugin-audioconvert.xml:
29936 * docs/plugins/inspect/plugin-audiorate.xml:
29937 * docs/plugins/inspect/plugin-audioresample.xml:
29938 * docs/plugins/inspect/plugin-audiotestsrc.xml:
29939 * docs/plugins/inspect/plugin-cdparanoia.xml:
29940 * docs/plugins/inspect/plugin-decodebin.xml:
29941 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
29942 * docs/plugins/inspect/plugin-gnomevfs.xml:
29943 * docs/plugins/inspect/plugin-libvisual.xml:
29944 * docs/plugins/inspect/plugin-ogg.xml:
29945 * docs/plugins/inspect/plugin-pango.xml:
29946 * docs/plugins/inspect/plugin-playbin.xml:
29947 * docs/plugins/inspect/plugin-subparse.xml:
29948 * docs/plugins/inspect/plugin-tcp.xml:
29949 * docs/plugins/inspect/plugin-theora.xml:
29950 * docs/plugins/inspect/plugin-typefindfunctions.xml:
29951 * docs/plugins/inspect/plugin-video4linux.xml:
29952 * docs/plugins/inspect/plugin-videorate.xml:
29953 * docs/plugins/inspect/plugin-videoscale.xml:
29954 * docs/plugins/inspect/plugin-videotestsrc.xml:
29955 * docs/plugins/inspect/plugin-volume.xml:
29956 * docs/plugins/inspect/plugin-vorbis.xml:
29957 * docs/plugins/inspect/plugin-ximagesink.xml:
29958 * docs/plugins/inspect/plugin-xvimagesink.xml:
29961 Original commit message from CVS:
29964 2006-04-28 10:42:57 +0000 Thomas Vander Stichele <thomas@apestaart.org>
29967 * win32/common/config.h:
29968 dist more win32 files
29969 Original commit message from CVS:
29970 dist more win32 files
29972 2006-04-28 10:41:57 +0000 Thomas Vander Stichele <thomas@apestaart.org>
29989 Original commit message from CVS:
29992 2006-04-27 00:19:29 +0000 David Schleef <ds@schleef.org>
29994 gst/videoscale/gstvideoscale.c: Add call to oil_init().
29995 Original commit message from CVS:
29996 * gst/videoscale/gstvideoscale.c: Add call to oil_init().
29999 2006-04-26 17:20:31 +0000 Thomas Vander Stichele <thomas@apestaart.org>
30003 * win32/common/config.h:
30005 Original commit message from CVS:
30008 2006-04-26 17:17:39 +0000 Wim Taymans <wim.taymans@gmail.com>
30010 ext/ogg/gstoggdemux.c: make sure correct newsegments are sent, so that the decoder and the demuxer agree on timestamp...
30011 Original commit message from CVS:
30012 2006-04-26 Thomas Vander Stichele <thomas at apestaart dot org>
30013 patch by: Wim Taymans
30014 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_packet),
30015 (gst_ogg_demux_perform_seek):
30016 make sure correct newsegments are sent, so that the decoder
30017 and the demuxer agree on timestamps. Fixes playback of a lot
30018 of Ogg files that do not start from 0. Fixes #339833.
30020 2006-04-26 16:44:20 +0000 Edward Hervey <edward@fluendo.com>
30022 Fix an infinite loop if frames are passed in with wrongly ordered timestamps. Fixes #339013.
30023 Original commit message from CVS:
30024 Patch by: Edward Hervey <edward@fluendo.com>
30025 * gst/videorate/gstvideorate.c: (gst_video_rate_chain):
30026 * tests/check/Makefile.am:
30027 * tests/check/elements/videorate.c: (assert_videorate_stats),
30028 (setup_videorate), (cleanup_videorate), (GST_START_TEST),
30029 (videorate_suite), (main):
30030 Fix an infinite loop if frames are passed in with wrongly ordered
30031 timestamps. Fixes #339013.
30033 2006-04-26 13:55:14 +0000 Thomas Vander Stichele <thomas@apestaart.org>
30036 * win32/common/config.h:
30038 Original commit message from CVS:
30041 2006-04-22 21:25:41 +0000 Tim-Philipp Müller <tim@centricular.net>
30043 gst/typefind/gsttypefindfunctions.c: fix typefinding on some ISO files. Fixes #339212.
30044 Original commit message from CVS:
30045 Patch by: Tim-Philipp Müller <tim at centricular dot net>
30046 * gst/typefind/gsttypefindfunctions.c: (qt_type_find):
30047 fix typefinding on some ISO files. Fixes #339212.
30049 2006-04-22 21:19:06 +0000 Tim-Philipp Müller <tim@centricular.net>
30051 gst-libs/gst/riff/riff-media.c: add another H264 fourcc. Fixes #339047.
30052 Original commit message from CVS:
30053 Patch by: Tim-Philipp Müller <tim at centricular dot net>
30054 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
30055 add another H264 fourcc. Fixes #339047.
30057 2006-04-22 21:12:52 +0000 Jan Schmidt <thaytan@mad.scientist.com>
30059 gst/playback/gststreamselector.c: Restore old StreamSelector behaviour.
30060 Original commit message from CVS:
30061 Patch by: Jan Schmidt
30062 * gst/playback/gststreamselector.c:
30063 (gst_stream_selector_bufferalloc):
30064 Restore old StreamSelector behaviour.
30067 2006-04-13 09:26:27 +0000 Thomas Vander Stichele <thomas@apestaart.org>
30070 * gst-libs/gst/rtp/Makefile.am:
30071 * gst-libs/gst/rtp/gstrtpbuffer.h:
30072 reverting rtp patches to fix freeze break on -base as explained on the list
30073 Original commit message from CVS:
30074 reverting rtp patches to fix freeze break on -base as explained on the list
30076 2006-04-13 03:55:12 +0000 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
30078 gst-libs/gst/rtp/gstrtpbuffer.h: Added GST_RTP_PAYLOAD_DYNAMIC_STRING for use by children
30079 Original commit message from CVS:
30080 2006-04-12 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
30081 * gst-libs/gst/rtp/gstrtpbuffer.h:
30082 Added GST_RTP_PAYLOAD_DYNAMIC_STRING for use by children
30083 * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
30084 * gst-libs/gst/rtp/gstbasertpaudiopayload.h:
30085 New RTP audio base payloader class. Supports frame or sample based codecs
30087 2006-04-12 11:04:53 +0000 Thomas Vander Stichele <thomas@apestaart.org>
30105 update libtool versioning
30106 Original commit message from CVS:
30107 update libtool versioning
30109 2006-04-12 10:58:00 +0000 Thomas Vander Stichele <thomas@apestaart.org>
30113 * win32/common/config.h:
30115 Original commit message from CVS:
30118 2006-04-11 17:31:29 +0000 Antoine Tremblay <hexa00@gmail.com>
30120 gst-libs/gst/rtp/gstbasertpdepayload.c: Fix some memory leaks: on finalize, free buffers left in the queue before des...
30121 Original commit message from CVS:
30122 Patch by: Antoine Tremblay <hexa00 at gmail dot com>
30123 * gst-libs/gst/rtp/gstbasertpdepayload.c:
30124 (gst_base_rtp_depayload_finalize), (gst_base_rtp_depayload_push):
30125 Fix some memory leaks: on finalize, free buffers left in the queue
30126 before destroying the queue; in _push(), unref rtp_buf even if
30127 the process vfunc returned a NULL buffer as output buffer (#337548);
30128 demote some recuring debug messages to LOG level.
30130 2006-04-11 15:01:51 +0000 Christian Schaller <uraeus@gnome.org>
30132 * gst-plugins-base.spec.in:
30133 fix version number macro
30134 Original commit message from CVS:
30135 fix version number macro
30137 2006-04-11 14:42:33 +0000 Wim Taymans <wim.taymans@gmail.com>
30139 ext/ogg/gstoggdemux.c: More cleanups.
30140 Original commit message from CVS:
30141 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_src_query),
30142 (gst_ogg_demux_chain_peer), (gst_ogg_pad_submit_packet),
30143 (gst_ogg_chain_free), (gst_ogg_demux_sink_event),
30144 (gst_ogg_demux_loop):
30146 Respect segment stop when emiting EOS or SEGMENT_DONE.
30149 2006-04-11 10:45:32 +0000 Tim-Philipp Müller <tim@centricular.net>
30151 gst/playback/gststreamselector.c: Don't leak pad name.
30152 Original commit message from CVS:
30153 * gst/playback/gststreamselector.c:
30154 (gst_stream_selector_get_property):
30155 Don't leak pad name.
30157 2006-04-11 09:42:52 +0000 Jan Schmidt <thaytan@mad.scientist.com>
30160 Mention bug #336617 closed by recent commit
30161 Original commit message from CVS:
30162 Mention bug #336617 closed by recent commit
30164 2006-04-10 20:32:46 +0000 Michael Smith <msmith@xiph.org>
30166 tests/check/: so that FC4 buildslaves can pass.
30167 Original commit message from CVS:
30168 * tests/check/Makefile.am:
30169 * tests/check/gst-plugins-base.supp:
30170 Suppress an old libtheora bug (fixed in more recent versions), so
30171 that FC4 buildslaves can pass.
30173 2006-04-10 19:13:30 +0000 Wim Taymans <wim.taymans@gmail.com>
30175 ext/ogg/gstoggdemux.c: Don't leak events.
30176 Original commit message from CVS:
30177 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_src_query),
30178 (gst_ogg_demux_receive_event), (gst_ogg_pad_event),
30179 (gst_ogg_demux_init), (gst_ogg_demux_finalize),
30180 (gst_ogg_demux_sink_event), (gst_ogg_demux_get_data),
30181 (gst_ogg_demux_loop):
30183 Remember what error we got when finding chains, if we
30184 were shutdown, that would not be an error.
30186 2006-04-10 17:05:46 +0000 Wim Taymans <wim.taymans@gmail.com>
30188 gst-libs/gst/audio/gstbaseaudiosink.c: Starting the ringbuffer when we did not acquire it can cause a deadlock, is po...
30189 Original commit message from CVS:
30190 * gst-libs/gst/audio/gstbaseaudiosink.c:
30191 (gst_base_audio_sink_event):
30192 Starting the ringbuffer when we did not acquire it can cause
30193 a deadlock, is pointless and causes nasty things for
30195 Fixes gst-launch audiotestsrc num-buffers=0 ! alsasink.
30197 2006-04-10 15:17:24 +0000 Wim Taymans <wim.taymans@gmail.com>
30199 ext/ogg/gstoggdemux.c: Add some more debugging.
30200 Original commit message from CVS:
30201 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_src_query),
30202 (gst_ogg_demux_receive_event), (gst_ogg_pad_event),
30203 (gst_ogg_demux_chain_peer), (gst_ogg_pad_submit_packet),
30204 (gst_ogg_demux_submit_buffer), (gst_ogg_demux_get_data),
30205 (gst_ogg_demux_deactivate_current_chain),
30206 (gst_ogg_demux_activate_chain), (gst_ogg_demux_perform_seek),
30207 (gst_ogg_demux_bisect_forward_serialno),
30208 (gst_ogg_demux_find_chains), (gst_ogg_demux_chain):
30209 Add some more debugging.
30211 2006-04-10 14:52:10 +0000 Thomas Vander Stichele <thomas@apestaart.org>
30214 * ext/theora/theoraenc.c:
30216 Original commit message from CVS:
30219 2006-04-10 10:29:21 +0000 Wim Taymans <wim.taymans@gmail.com>
30221 ext/theora/theoradec.c: Some more debug info.
30222 Original commit message from CVS:
30223 * ext/theora/theoradec.c: (theora_dec_src_event),
30224 (theora_handle_data_packet):
30225 Some more debug info.
30226 * tests/examples/seek/seek.c: (start_seek), (main):
30227 Print element messages too.
30229 2006-04-09 17:14:22 +0000 Sébastien Moutte <sebastien@moutte.net>
30231 gst/audioresample/debug.h: replace debug macros with variable number of parameters by a simple alias to gstreamer sta...
30232 Original commit message from CVS:
30233 * gst/audioresample/debug.h:
30234 replace debug macros with variable number of parameters
30235 by a simple alias to gstreamer standard debug macros
30236 (#define RESAMPLE_ERROR GST_ERROR, __VA_ARGS__ is not
30237 supported by MSVC 6.0 and 7.1)
30238 * gst/audioresample/resample.h:
30239 define M_PI and rint for WIN32
30240 * win32/common/libgstaudio.def:
30241 * win32/common/libgstriff.def:
30242 * win32/common/libgsttag.def:
30243 * win32/common/libgstvideo.def:
30244 add new exported functions
30246 update project files
30248 2006-04-08 21:02:53 +0000 Stefan Kost <ensonic@users.sourceforge.net>
30250 Fix #337365 (g_type_class_ref <-> g_type_class_peek_parent)
30251 Original commit message from CVS:
30252 * ext/alsa/gstalsamixeroptions.c:
30253 (gst_alsa_mixer_options_class_init):
30254 * ext/alsa/gstalsamixertrack.c: (gst_alsa_mixer_track_class_init):
30255 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_class_init):
30256 * ext/ogg/gstoggmux.c: (gst_ogg_mux_class_init):
30257 * ext/ogg/gstoggparse.c: (gst_ogg_parse_class_init):
30258 * gst-libs/gst/audio/gstaudioclock.c: (gst_audio_clock_class_init):
30259 * gst-libs/gst/audio/gstaudiofilter.c:
30260 (gst_audio_filter_class_init):
30261 * gst-libs/gst/audio/gstaudiosink.c:
30262 (gst_audioringbuffer_class_init):
30263 * gst-libs/gst/audio/gstaudiosrc.c:
30264 (gst_audioringbuffer_class_init):
30265 * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_class_init):
30266 * gst-libs/gst/interfaces/colorbalancechannel.c:
30267 (gst_color_balance_channel_class_init):
30268 * gst-libs/gst/interfaces/mixeroptions.c:
30269 (gst_mixer_options_class_init):
30270 * gst-libs/gst/interfaces/mixertrack.c:
30271 (gst_mixer_track_class_init):
30272 * gst-libs/gst/interfaces/tunerchannel.c:
30273 (gst_tuner_channel_class_init):
30274 * gst-libs/gst/interfaces/tunernorm.c: (gst_tuner_norm_class_init):
30275 * gst-libs/gst/netbuffer/gstnetbuffer.c:
30276 (gst_netbuffer_class_init):
30277 * gst-libs/gst/rtp/gstbasertppayload.c:
30278 (gst_basertppayload_class_init):
30279 * gst/playback/gstdecodebin.c: (gst_decode_bin_class_init):
30280 * gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init):
30281 * gst/playback/gstplaybin.c: (gst_play_bin_class_init):
30282 * gst/playback/gststreaminfo.c: (gst_stream_info_class_init):
30283 * gst/playback/gststreamselector.c:
30284 (gst_stream_selector_class_init):
30285 * gst/subparse/gstsubparse.c: (gst_sub_parse_class_init):
30286 * gst/tcp/gsttcpclientsink.c: (gst_tcp_client_sink_class_init):
30287 * sys/v4l/gstv4lcolorbalance.c:
30288 (gst_v4l_color_balance_channel_class_init):
30289 * sys/v4l/gstv4ljpegsrc.c: (gst_v4ljpegsrc_class_init):
30290 * sys/v4l/gstv4lmjpegsink.c: (gst_v4lmjpegsink_class_init):
30291 * sys/v4l/gstv4lmjpegsrc.c: (gst_v4lmjpegsrc_class_init):
30292 * sys/v4l/gstv4ltuner.c: (gst_v4l_tuner_channel_class_init),
30293 (gst_v4l_tuner_norm_class_init):
30294 * sys/ximage/ximagesink.c: (gst_ximagesink_class_init):
30295 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_class_init):
30296 * tests/old/testsuite/alsa/sinesrc.c: (sinesrc_class_init):
30297 Fix #337365 (g_type_class_ref <-> g_type_class_peek_parent)
30299 2006-04-08 18:09:17 +0000 Stefan Kost <ensonic@users.sourceforge.net>
30301 Fix broken GObject macros
30302 Original commit message from CVS:
30303 * ext/pango/gsttextrender.h:
30304 * gst-libs/gst/audio/gstaudiosink.h:
30305 * gst-libs/gst/audio/gstaudiosrc.h:
30306 * gst-libs/gst/audio/gstbaseaudiosink.h:
30307 * gst-libs/gst/audio/gstbaseaudiosrc.h:
30308 * gst-libs/gst/audio/gstringbuffer.h:
30309 * gst-libs/gst/rtp/gstbasertpdepayload.h:
30310 * gst-libs/gst/rtp/gstbasertppayload.h:
30311 * gst-libs/gst/video/gstvideofilter.h:
30312 * gst-libs/gst/video/gstvideosink.h:
30313 * gst/playback/gstplaybasebin.h:
30314 * gst/tcp/gstmultifdsink.h:
30315 * sys/v4l/gstv4lelement.h:
30316 Fix broken GObject macros
30318 2006-04-08 16:21:15 +0000 Stefan Kost <ensonic@users.sourceforge.net>
30320 ext/alsa/gstalsasink.c: More debug to trace why my USB headset is not working with gst
30321 Original commit message from CVS:
30322 * ext/alsa/gstalsasink.c: (set_hwparams), (alsasink_parse_spec):
30323 More debug to trace why my USB headset is not working with gst
30325 2006-04-07 17:18:11 +0000 Jan Schmidt <thaytan@mad.scientist.com>
30327 gst/playback/gstplaybasebin.c: Clean up our group elements properly in the case where it never got committed - it sti...
30328 Original commit message from CVS:
30329 * gst/playback/gstplaybasebin.c: (group_destroy):
30330 Clean up our group elements properly in the case where it never
30331 got committed - it still got added unconditionally to the bin.
30333 2006-04-07 15:14:32 +0000 Wim Taymans <wim.taymans@gmail.com>
30335 ext/theora/theoradec.c: Unref unhandled events.
30336 Original commit message from CVS:
30337 * ext/theora/theoradec.c: (theora_dec_sink_event),
30338 (theora_handle_data_packet), (theora_dec_chain):
30339 Unref unhandled events.
30340 Protect against empty buffers.
30341 Perform QoS on running time.
30343 2006-04-07 13:24:54 +0000 Michael Smith <msmith@xiph.org>
30345 ext/vorbis/vorbisenc.c: Remove leaks from vorbisenc.
30346 Original commit message from CVS:
30347 * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_set_header_on_caps),
30348 (gst_vorbis_enc_chain):
30349 Remove leaks from vorbisenc.
30350 Mostly minor changes, the only significant one is that now the
30351 buffers we set as 'streamheader' on the caps are copies of the
30352 original buffers, to avoid circular refcounting problems.
30354 2006-04-07 09:51:35 +0000 Jan Schmidt <thaytan@mad.scientist.com>
30356 gst/playback/gstplaybasebin.c: Don't remove our mute-probe if someone else already did so.
30357 Original commit message from CVS:
30358 * gst/playback/gstplaybasebin.c: (mute_stream), (setup_substreams):
30359 Don't remove our mute-probe if someone else already did so.
30360 Don't set a 2nd one if there is already one pending on the pad.
30361 * gst/playback/gstplaybin.c: (gst_play_bin_send_event_to_sink),
30363 When a seek fails, ensure that playbin is still set back to playing.
30364 * gst/typefind/gsttypefindfunctions.c: (mpeg_ts_probe_headers),
30365 (mpeg_ts_type_find), (plugin_init):
30366 Add a typefind function for mpeg-ts streams.
30368 2006-04-06 11:40:45 +0000 Andy Wingo <wingo@pobox.com>
30371 * gst/audiotestsrc/gstaudiotestsrc.c:
30372 * gst/videorate/gstvideorate.c:
30373 gst/videorate/gstvideorate.c (gst_video_rate_reset)
30374 Original commit message from CVS:
30375 2006-04-06 Andy Wingo <wingo@pobox.com>
30376 * gst/videorate/gstvideorate.c (gst_video_rate_reset)
30377 (gst_video_rate_init): Caps-related parameters should not be reset
30378 by a flush -- move their inits to the instance init function.
30379 (gst_video_rate_flush_prev): Don't complain if gst_pad_push
30380 is not OK, just return the result.
30381 * gst/audiotestsrc/gstaudiotestsrc.c
30382 (gst_audio_test_src_class_init)
30383 (gst_audio_test_src_get_times): Re-enable is-live=true, as was
30384 broken by Stefan's commit on 24 March.
30386 2006-04-06 10:50:14 +0000 Andy Wingo <wingo@pobox.com>
30388 ext/ogg/gstoggmux.c (gst_ogg_mux_push_buffer): Set caps on buffers being pushed out. Fixes oggmux ! multifdsink.
30389 Original commit message from CVS:
30390 2006-04-06 Andy Wingo <wingo@pobox.com>
30391 * ext/ogg/gstoggmux.c (gst_ogg_mux_push_buffer): Set caps on
30392 buffers being pushed out. Fixes oggmux ! multifdsink.
30394 2006-04-05 13:05:25 +0000 Tim-Philipp Müller <tim@centricular.net>
30396 ext/vorbis/: Remove left-over 0.8 cruft; use GST_DEBUG_FUNCPTR; make vorbisenc adhere to the official nomenclature; u...
30397 Original commit message from CVS:
30398 * ext/vorbis/vorbisdec.c: (gst_vorbis_dec_class_init),
30399 (gst_vorbis_dec_init), (vorbis_dec_finalize):
30400 * ext/vorbis/vorbisdec.h:
30401 * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_add_interfaces),
30402 (gst_vorbis_enc_base_init), (gst_vorbis_enc_class_init),
30403 (gst_vorbis_enc_sink_setcaps), (gst_vorbis_enc_convert_src),
30404 (gst_vorbis_enc_convert_sink), (gst_vorbis_enc_get_query_types),
30405 (gst_vorbis_enc_src_query), (gst_vorbis_enc_sink_query),
30406 (gst_vorbis_enc_init), (gst_vorbis_enc_get_tag_value),
30407 (gst_vorbis_enc_metadata_set1), (gst_vorbis_enc_set_metadata),
30408 (gst_vorbis_enc_setup), (gst_vorbis_enc_clear),
30409 (gst_vorbis_enc_buffer_from_packet),
30410 (gst_vorbis_enc_buffer_from_header_packet),
30411 (gst_vorbis_enc_push_buffer), (gst_vorbis_enc_push_packet),
30412 (gst_vorbis_enc_set_header_on_caps), (gst_vorbis_enc_sink_event),
30413 (gst_vorbis_enc_chain), (gst_vorbis_enc_output_buffers),
30414 (gst_vorbis_enc_get_property), (gst_vorbis_enc_set_property),
30415 (gst_vorbis_enc_change_state):
30416 * ext/vorbis/vorbisenc.h:
30417 Remove left-over 0.8 cruft; use GST_DEBUG_FUNCPTR; make
30418 vorbisenc adhere to the official nomenclature; use boilerplate
30421 2006-04-04 11:20:24 +0000 Andy Wingo <wingo@pobox.com>
30423 gst/videorate/gstvideorate.c (gst_video_rate_flush_prev): Whoops, fix bug introduced. Bad hacker!
30424 Original commit message from CVS:
30425 2006-04-04 Andy Wingo <wingo@pobox.com>
30426 * gst/videorate/gstvideorate.c (gst_video_rate_flush_prev):
30427 Whoops, fix bug introduced. Bad hacker!
30429 2006-04-04 11:15:00 +0000 Andy Wingo <wingo@pobox.com>
30431 gst/videorate/gstvideorate.c (gst_video_rate_flush_prev): Properly handle the case where you get EOS before any buffe...
30432 Original commit message from CVS:
30433 2006-04-04 Andy Wingo <wingo@pobox.com>
30434 * gst/videorate/gstvideorate.c (gst_video_rate_flush_prev):
30435 Properly handle the case where you get EOS before any buffers are
30436 received. Use gst_buffer_make_metadata_writable where appropriate.
30438 2006-04-04 10:16:46 +0000 Andy Wingo <wingo@pobox.com>
30440 ext/theora/theoradec.c (theora_handle_data_packet): This value is often negative -- make it signed so as not to wrap ...
30441 Original commit message from CVS:
30442 2006-04-04 Andy Wingo <wingo@pobox.com>
30443 * ext/theora/theoradec.c (theora_handle_data_packet): This value
30444 is often negative -- make it signed so as not to wrap around.
30445 Fixes segfaults introduced on 9 March.
30447 2006-04-03 16:43:10 +0000 Wim Taymans <wim.taymans@gmail.com>
30449 ext/theora/: Don't try to store a gdouble in a gboolean.
30450 Original commit message from CVS:
30451 * ext/theora/gsttheoradec.h:
30452 * ext/theora/theoradec.c: (theora_dec_src_event):
30453 Don't try to store a gdouble in a gboolean.
30456 2006-04-03 12:55:18 +0000 Michael Smith <msmith@xiph.org>
30458 ext/ogg/gstoggmux.c: Oggmux sucks.
30459 Original commit message from CVS:
30460 * ext/ogg/gstoggmux.c: (gst_ogg_mux_queue_pads):
30462 Make it suck slightly less by writing out the final page.
30463 Still can't encode a vorbis-in-ogg file correctly, though.
30465 2006-04-03 08:49:06 +0000 Andy Wingo <wingo@pobox.com>
30467 ext/theora/theoraparse.c (theora_parse_drain_queue): Um, remove a g_print.
30468 Original commit message from CVS:
30469 2006-04-03 Andy Wingo <wingo@pobox.com>
30470 * ext/theora/theoraparse.c (theora_parse_drain_queue): Um, remove
30473 2006-04-03 08:32:21 +0000 Andy Wingo <wingo@pobox.com>
30475 ext/theora/theora.c (plugin_init): Register theoraparse.
30476 Original commit message from CVS:
30477 2006-04-03 Andy Wingo <wingo@pobox.com>
30478 * ext/theora/theora.c (plugin_init): Register theoraparse.
30479 * ext/theora/gsttheoraparse.h:
30480 * ext/theora/theoraparse.c: New files implementing a theora
30481 parser. Now we can properly remux ogg/theora+vorbis, yay.
30483 2006-04-03 08:28:58 +0000 Andy Wingo <wingo@pobox.com>
30485 ext/vorbis/vorbisparse.c: Add some docs and a copyright.
30486 Original commit message from CVS:
30487 2006-04-03 Andy Wingo <wingo@pobox.com>
30488 * ext/vorbis/vorbisparse.c: Add some docs and a copyright.
30490 2006-04-01 15:34:38 +0000 Thomas Vander Stichele <thomas@apestaart.org>
30494 don't use AS_LIBTOOL_TAGS, it doesn't work
30495 Original commit message from CVS:
30496 don't use AS_LIBTOOL_TAGS, it doesn't work
30498 2006-04-01 11:41:13 +0000 Thomas Vander Stichele <thomas@apestaart.org>
30501 * ext/pango/gsttextoverlay.c:
30502 * sys/v4l/gstv4lsrc.c:
30503 remove BT8x8 from description, works for more devices
30504 Original commit message from CVS:
30505 remove BT8x8 from description, works for more devices
30507 2006-04-01 11:21:30 +0000 Thomas Vander Stichele <thomas@apestaart.org>
30509 gst/audiotestsrc/gstaudiotestsrc.c: Fixed the sample pipeline (see #323798)
30510 Original commit message from CVS:
30511 * gst/audiotestsrc/gstaudiotestsrc.c:
30512 Fixed the sample pipeline (see #323798)
30514 2006-04-01 09:50:34 +0000 Thomas Vander Stichele <thomas@apestaart.org>
30516 use AS_VERSION and AS_NANO more cleanups
30517 Original commit message from CVS:
30519 * win32/common/config.h:
30520 * win32/common/config.h.in:
30521 use AS_VERSION and AS_NANO
30524 2006-03-31 17:08:41 +0000 Andy Wingo <wingo@pobox.com>
30526 ext/vorbis/vorbisparse.c (vorbis_parse_sink_event): Fix uninitialized variable return that would happen.
30527 Original commit message from CVS:
30528 2006-03-31 Andy Wingo <wingo@pobox.com>
30529 * ext/vorbis/vorbisparse.c (vorbis_parse_sink_event): Fix
30530 uninitialized variable return that would happen.
30532 2006-03-31 16:57:47 +0000 Andy Wingo <wingo@pobox.com>
30534 ext/vorbis/vorbisparse.c (vorbis_parse_drain_queue): Fix uninitialized variable return that would never happen.
30535 Original commit message from CVS:
30536 2006-03-31 Andy Wingo <wingo@pobox.com>
30537 * ext/vorbis/vorbisparse.c (vorbis_parse_drain_queue): Fix
30538 uninitialized variable return that would never happen.
30540 2006-03-31 16:43:43 +0000 Andy Wingo <wingo@pobox.com>
30542 ext/vorbis/vorbisparse.c (gst_vorbis_parse_init)
30543 Original commit message from CVS:
30544 2006-03-31 Andy Wingo <wingo@pobox.com>
30545 * ext/vorbis/vorbisparse.c (gst_vorbis_parse_init)
30546 (vorbis_parse_sink_event): Add an event function to flush our
30547 state on a seek, and to drain buffers on a premature EOS.
30548 (vorbis_parse_push_headers, vorbis_parse_clear_queue)
30549 (vorbis_parse_push_buffer, vorbis_parse_drain_queue_prematurely)
30550 (vorbis_parse_chain, vorbis_parse_queue_buffer)
30551 (vorbis_parse_drain_queue): Queue up buffers until we can set
30552 their timestamps and granulepos values.
30553 * ext/vorbis/vorbisparse.h: Include the vorbis decoder headers,
30554 and keep track of data needed for deriving granulepos and
30555 timestamps for buffers.
30557 2006-03-30 11:05:25 +0000 Thomas Vander Stichele <thomas@apestaart.org>
30559 * pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
30560 * pkgconfig/gstreamer-plugins-base.pc.in:
30561 expose pluginsdir so gonlin can use it for tests
30562 Original commit message from CVS:
30563 expose pluginsdir so gonlin can use it for tests
30565 2006-03-30 10:03:56 +0000 Thomas Vander Stichele <thomas@apestaart.org>
30567 * pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
30568 * pkgconfig/gstreamer-plugins-base.pc.in:
30569 add ccda to libraries
30570 Original commit message from CVS:
30571 add ccda to libraries
30573 2006-03-29 14:00:08 +0000 j^ <j@bootlab.org>
30575 better/unified long descriptions
30576 Original commit message from CVS:
30577 Patch by: j^ <j at bootlab dot org>
30578 * ext/alsa/gstalsamixerelement.c:
30579 (gst_alsa_mixer_element_class_init):
30580 * ext/alsa/gstalsasink.c:
30581 * ext/alsa/gstalsasrc.c: (gst_alsasrc_class_init):
30582 * ext/ogg/gstoggdemux.c:
30583 * ext/ogg/gstoggmux.c:
30584 * ext/ogg/gstoggparse.c:
30585 * ext/pango/gstclockoverlay.c:
30586 * ext/pango/gsttextoverlay.c:
30587 * ext/pango/gsttextrender.c:
30588 * ext/pango/gsttimeoverlay.c:
30589 * ext/theora/theoradec.c:
30590 * ext/theora/theoraenc.c:
30591 * ext/vorbis/vorbisdec.c:
30592 * ext/vorbis/vorbisenc.c:
30593 * gst/audioconvert/gstaudioconvert.c:
30594 * gst/subparse/gstsubparse.c:
30595 * gst/tcp/gstmultifdsink.c:
30596 * gst/tcp/gsttcpclientsink.c:
30597 * gst/tcp/gsttcpclientsrc.c:
30598 * gst/tcp/gsttcpserversink.c:
30599 * gst/tcp/gsttcpserversrc.c:
30600 better/unified long descriptions
30603 2006-03-29 13:54:24 +0000 Wim Taymans <wim.taymans@gmail.com>
30605 tests/examples/seek/seek.c: Don't let double and tripple clicks mess up our state.
30606 Original commit message from CVS:
30607 * tests/examples/seek/seek.c: (end_scrub), (seek_cb), (start_seek),
30609 Don't let double and tripple clicks mess up our state.
30611 2006-03-28 13:13:43 +0000 Tim-Philipp Müller <tim@centricular.net>
30613 gst/playback/gstplaybin.c: Error out gracefully when we can't create any of the usual conversion elements for some re...
30614 Original commit message from CVS:
30615 * gst/playback/gstplaybin.c: (gen_video_element),
30616 (gen_text_element), (gen_audio_element), (gen_vis_element):
30617 Error out gracefully when we can't create any of the usual
30618 conversion elements for some reason. Also, don't try to
30619 create an audioscale (sic) element that's not used anyway.
30621 2006-03-28 10:21:52 +0000 Tim-Philipp Müller <tim@centricular.net>
30623 gst/playback/gstplaybasebin.c: Don't post RESOURCE_NOT_FOUND error when we can't find a source element for a particul...
30624 Original commit message from CVS:
30625 * gst/playback/gstplaybasebin.c: (setup_source):
30626 Don't post RESOURCE_NOT_FOUND error when we can't find a source
30627 element for a particular protocol, that's confusing for users.
30628 Instead, post a RESOURCE_FAILED error, so that our own error
30629 message is actually shown in totem etc. (#336303).
30631 2006-03-27 16:36:46 +0000 Tommi Myöhänen <ext-tommi.myohanen@nokia.com>
30633 ext/gnomevfs/gstgnomevfssrc.c: Fix some minor memory leaks (#336194).
30634 Original commit message from CVS:
30635 Patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com>
30636 * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_finalize),
30637 (gst_gnome_vfs_src_get_icy_metadata):
30638 Fix some minor memory leaks (#336194).
30640 2006-03-27 16:15:00 +0000 Tim-Philipp Müller <tim@centricular.net>
30642 ext/gnomevfs/: Make gnomevfssink accept filenames as well as URIs for the "location" property, just like gnomevfssrc ...
30643 Original commit message from CVS:
30644 * ext/gnomevfs/gstgnomevfs.c:
30645 (gst_gnome_vfs_location_to_uri_string):
30646 * ext/gnomevfs/gstgnomevfs.h:
30647 * ext/gnomevfs/gstgnomevfssink.c:
30648 (gst_gnome_vfs_sink_set_property):
30649 * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_set_property):
30650 Make gnomevfssink accept filenames as well as URIs for the
30651 "location" property, just like gnomevfssrc does (and
30652 filesrc/filesink do) (#336190).
30654 2006-03-24 20:35:34 +0000 Thomas Vander Stichele <thomas@apestaart.org>
30656 tests/check/generic/clock-selection.c: set to NULL before unreffing, fixes a valgrind leak.
30657 Original commit message from CVS:
30658 * tests/check/generic/clock-selection.c: (GST_START_TEST):
30659 set to NULL before unreffing, fixes a valgrind leak.
30660 Why was this not triggering the error that an object needs to
30661 be NULL before unreffing ?
30662 * win32/common/config.h:
30665 2006-03-24 17:57:39 +0000 Tim-Philipp Müller <tim@centricular.net>
30667 gst/subparse/gstsubparse.*: Text subtitle files may or may not be UTF-8. If it's not, we don't really want to see '?'...
30668 Original commit message from CVS:
30669 * gst/subparse/gstsubparse.c: (convert_encoding),
30670 (gst_sub_parse_change_state):
30671 * gst/subparse/gstsubparse.h:
30672 Text subtitle files may or may not be UTF-8. If it's not, we
30673 don't really want to see '?' characters in place of non-ASCII
30674 characters like accented characters. So let's assume the input
30675 is UTF-8 until we come across text that is clearly not. If it's
30676 not UTF-8, we don't really know what it is, so try the following:
30677 (a) see whether the GST_SUBTITLE_ENCODING environment variable
30678 is set; if not, check (b) if the current locale encoding is
30679 non-UTF-8 and use that if it is, or (c) assume ISO-8859-15 if
30680 the current locale encoding is UTF-8 and the environment variable
30681 was not set to any particular encoding. Not perfect, but better
30682 than nothing (and better than before, I think) (fixes #172848).
30684 2006-03-24 17:39:45 +0000 Thomas Vander Stichele <thomas@apestaart.org>
30686 * docs/plugins/tmpl/.gitignore:
30687 * tests/check/libs/.gitignore:
30688 * tests/check/pipelines/.gitignore:
30689 * tests/examples/volume/.gitignore:
30691 Original commit message from CVS:
30694 2006-03-24 17:26:54 +0000 Thomas Vander Stichele <thomas@apestaart.org>
30696 configure.ac: update core requirement to 0.10.4.1 because of async_playback vmethod on GstBaseSink
30697 Original commit message from CVS:
30698 2006-03-24 Thomas Vander Stichele <thomas at apestaart dot org>
30700 update core requirement to 0.10.4.1 because of async_playback
30701 vmethod on GstBaseSink
30703 2006-03-24 17:11:53 +0000 Stefan Kost <ensonic@users.sourceforge.net>
30705 use DEBUG_FUNCPTR for collectpads
30706 Original commit message from CVS:
30707 * ext/ogg/gstoggmux.c: (gst_ogg_mux_init):
30708 * gst/adder/gstadder.c: (gst_adder_init):
30709 use DEBUG_FUNCPTR for collectpads
30711 2006-03-24 14:11:20 +0000 Thomas Vander Stichele <thomas@apestaart.org>
30714 don't go through check-torture if no check installed
30715 Original commit message from CVS:
30716 don't go through check-torture if no check installed
30718 2006-03-24 10:42:11 +0000 Stefan Kost <ensonic@users.sourceforge.net>
30720 Add docs for adder, use GST_ELEMENT_DETAILS macro, define GstElementDetails at the top
30721 Original commit message from CVS:
30722 * docs/plugins/Makefile.am:
30723 * docs/plugins/gst-plugins-base-plugins-docs.sgml:
30724 * docs/plugins/gst-plugins-base-plugins-sections.txt:
30725 * ext/cdparanoia/gstcdparanoiasrc.c:
30726 * ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_base_init),
30727 (gst_gnome_vfs_sink_class_init):
30728 * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_base_init):
30729 * ext/ogg/gstoggdemux.c: (gst_ogg_demux_base_init):
30730 * ext/ogg/gstoggmux.c:
30731 * ext/ogg/gstoggparse.c: (gst_ogg_parse_base_init):
30732 * ext/ogg/gstogmparse.c: (gst_ogm_audio_parse_base_init),
30733 (gst_ogm_video_parse_base_init), (gst_ogm_text_parse_base_init):
30734 * ext/pango/gsttextoverlay.c:
30735 * ext/pango/gsttextrender.c:
30736 * ext/theora/theoradec.c:
30737 * ext/theora/theoraenc.c:
30738 * ext/vorbis/vorbisdec.c:
30739 * ext/vorbis/vorbisenc.c:
30740 * gst-libs/gst/audio/gstaudiofilter.c:
30741 (gst_audio_filter_base_init):
30742 * gst-libs/gst/audio/gstaudiofiltertemplate.c:
30743 (gst_audio_filter_template_base_init):
30744 * gst/adder/gstadder.c: (gst_adder_get_type):
30745 * gst/adder/gstadder.h:
30746 * gst/audioconvert/gstaudioconvert.c:
30747 * gst/audiotestsrc/gstaudiotestsrc.c:
30748 (gst_audiostestsrc_wave_get_type), (gst_audio_test_src_class_init),
30749 (gst_audio_test_src_create):
30750 * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
30751 * gst/playback/gstdecodebin.c:
30752 * gst/playback/gstplaybin.c:
30753 * gst/playback/gststreamselector.c:
30754 (gst_stream_selector_base_init):
30755 * gst/subparse/gstsubparse.c: (gst_sub_parse_base_init):
30756 * gst/volume/gstvolume.c:
30757 * sys/v4l/gstv4lmjpegsink.c:
30758 * sys/v4l/gstv4lmjpegsrc.c:
30759 * tests/check/libs/cddabasesrc.c:
30760 * tests/old/examples/gob/gst-identity2.gob:
30761 Add docs for adder, use GST_ELEMENT_DETAILS macro,
30762 define GstElementDetails at the top
30764 2006-03-23 21:48:18 +0000 Sébastien Moutte <sebastien@moutte.net>
30766 win32/common/libgstinterfaces.def: Add a lot of export functions for gst-python
30767 Original commit message from CVS:
30768 * win32/common/libgstinterfaces.def:
30769 Add a lot of export functions for gst-python
30770 * win32/common/libgstinterfaces.dsp:
30771 Add a missing include folder in the project configuration
30773 2006-03-23 16:58:03 +0000 Wim Taymans <wim.taymans@gmail.com>
30775 gst-libs/gst/audio/gstbaseaudiosrc.c: Fix audio sources, forgot to make the ringbuffer startable...
30776 Original commit message from CVS:
30777 * gst-libs/gst/audio/gstbaseaudiosrc.c:
30778 (gst_base_audio_src_get_time), (gst_base_audio_src_create),
30779 (gst_base_audio_src_change_state):
30780 Fix audio sources, forgot to make the ringbuffer
30783 2006-03-23 16:29:58 +0000 Wim Taymans <wim.taymans@gmail.com>
30785 gst-libs/gst/audio/gstbaseaudiosrc.c: unparent instead of unref the ringbuffer.
30786 Original commit message from CVS:
30787 * gst-libs/gst/audio/gstbaseaudiosrc.c:
30788 (gst_base_audio_src_get_time), (gst_base_audio_src_create),
30789 (gst_base_audio_src_change_state):
30790 unparent instead of unref the ringbuffer.
30792 2006-03-23 16:24:23 +0000 Wim Taymans <wim.taymans@gmail.com>
30794 gst-libs/gst/audio/gstbaseaudiosink.c: Implement new async_play vmethod to start slaving and allow playback start in ...
30795 Original commit message from CVS:
30796 * gst-libs/gst/audio/gstbaseaudiosink.c:
30797 (gst_base_audio_sink_class_init), (gst_base_audio_sink_async_play),
30798 (gst_base_audio_sink_do_play), (gst_base_audio_sink_change_state):
30799 Implement new async_play vmethod to start slaving and allow
30800 playback start in case of async PLAY state changes.
30801 * gst-libs/gst/video/gstvideosink.c: (gst_video_sink_init):
30802 Enable QoS with new method in base class.
30804 2006-03-23 11:18:19 +0000 Julien MOUTTE <julien@moutte.net>
30806 gst/videotestsrc/gstvideotestsrc.c: Partially handle 0 framerate, only EOS after the first frame is missing.
30807 Original commit message from CVS:
30808 Patch by: Julien MOUTTE <julien at moutte dot net>
30809 * gst/videotestsrc/gstvideotestsrc.c: (gst_video_test_src_query),
30810 (gst_video_test_src_do_seek), (gst_video_test_src_create):
30811 Partially handle 0 framerate, only EOS after the first frame
30814 2006-03-23 09:38:59 +0000 Fabrizio Gennari <fabrizio.ge@tiscali.it>
30816 gst/: Patch for support of YVU9 AVI files (#334822)
30817 Original commit message from CVS:
30818 Patch by: Fabrizio Gennari <fabrizio dot ge at tiscali dot it>
30819 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
30820 (gst_riff_create_video_template_caps):
30821 * gst/ffmpegcolorspace/avcodec.h:
30822 * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
30823 (gst_ffmpeg_pixfmt_to_caps), (gst_ffmpeg_caps_to_pixfmt),
30824 (gst_ffmpegcsp_avpicture_fill):
30825 * gst/ffmpegcolorspace/imgconvert.c:
30826 Patch for support of YVU9 AVI files (#334822)
30828 2006-03-22 15:29:25 +0000 Edward Hervey <bilboed@bilboed.com>
30830 docs/design/design-decodebin.txt: Added design document for new decodebin text/x-pango-markup is also a default targe...
30831 Original commit message from CVS:
30832 * docs/design/design-decodebin.txt:
30833 Added design document for new decodebin
30834 (Target Caps): text/x-pango-markup is also a default target caps.
30836 2006-03-22 15:11:47 +0000 Edward Hervey <bilboed@bilboed.com>
30838 docs/design/design-decodebin.txt: Added design document for new decodebin
30839 Original commit message from CVS:
30840 * docs/design/design-decodebin.txt:
30841 Added design document for new decodebin
30843 2006-03-22 12:33:09 +0000 Wim Taymans <wim.taymans@gmail.com>
30845 gst-libs/gst/audio/gstbaseaudiosink.c: Since we _parent the ringbuffer, we also need to _unparent instead of a plain ...
30846 Original commit message from CVS:
30847 * gst-libs/gst/audio/gstbaseaudiosink.c:
30848 (gst_base_audio_sink_dispose):
30849 Since we _parent the ringbuffer, we also need to
30850 _unparent instead of a plain _unref.
30852 2006-03-22 12:28:36 +0000 Wim Taymans <wim.taymans@gmail.com>
30854 tests/examples/seek/seek.c: Add scrub checkbox.
30855 Original commit message from CVS:
30856 * tests/examples/seek/seek.c: (end_scrub), (do_seek), (seek_cb),
30857 (stop_seek), (scrub_toggle_cb), (main):
30858 Add scrub checkbox.
30860 2006-03-21 17:47:04 +0000 Tim-Philipp Müller <tim@centricular.net>
30862 ext/ogg/gstoggparse.c: Fix very inefficient usage of linked lists (#335365).
30863 Original commit message from CVS:
30864 * ext/ogg/gstoggparse.c: (gst_ogg_parse_find_stream),
30865 (gst_ogg_parse_chain):
30866 Fix very inefficient usage of linked lists (#335365).
30868 2006-03-21 14:26:01 +0000 Edward Hervey <bilboed@bilboed.com>
30870 gcc 4.1 unreferenced pointer fixes.
30871 Original commit message from CVS:
30872 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_dispose):
30873 * gst/playback/gstplaybin.c: (handoff):
30874 * gst/playback/gststreamselector.c:
30875 (gst_stream_selector_set_property):
30876 gcc 4.1 unreferenced pointer fixes.
30877 * sys/ximage/ximagesink.c: (gst_ximagesink_ximage_put):
30878 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_put):
30879 gst_buffer_ref() now takes a GstBuffer*.
30881 2006-03-20 18:09:41 +0000 Julien Moutte <julien@moutte.net>
30883 sys/xvimage/xvimagesink.c: Fix a memleak reported by Jan Schmidt.
30884 Original commit message from CVS:
30885 2006-03-20 Julien MOUTTE <julien@moutte.net>
30886 * sys/xvimage/xvimagesink.c:
30887 (gst_xvimagesink_get_format_from_caps): Fix a memleak reported
30890 2006-03-19 11:37:46 +0000 Tim-Philipp Müller <tim@centricular.net>
30892 gst/typefind/gsttypefindfunctions.c: Can't do tag preferences via probability, as tags would then lose against types ...
30893 Original commit message from CVS:
30894 * gst/typefind/gsttypefindfunctions.c: (id3v2_type_find),
30895 (id3v1_type_find), (apetag_type_find), (plugin_init):
30896 Can't do tag preferences via probability, as tags would then
30897 lose against types that are recognised with MAXIMUM probability
30898 (like .wav); so let all tag typefinders return MAXIMUM themselves
30899 and order them via the rank. Split ID3v1 and ID3v2 typefinders so
30900 that we can prefer APE to ID3v1 (fixes #335028).
30902 2006-03-17 17:48:33 +0000 Wim Taymans <wim.taymans@gmail.com>
30904 gst-libs/gst/audio/: Only start playback if we are playing. should fix #330748.
30905 Original commit message from CVS:
30906 * gst-libs/gst/audio/gstbaseaudiosink.c:
30907 (gst_base_audio_sink_change_state):
30908 * gst-libs/gst/audio/gstringbuffer.c: (wait_segment),
30909 (gst_ring_buffer_may_start):
30910 * gst-libs/gst/audio/gstringbuffer.h:
30911 Only start playback if we are playing.
30912 should fix #330748.
30914 2006-03-17 13:11:45 +0000 Jan Schmidt <thaytan@mad.scientist.com>
30916 Revert accidental commits to these files.
30917 Original commit message from CVS:
30918 * ext/alsa/gstalsasink.c: (gst_alsasink_getcaps):
30919 * win32/common/config.h:
30920 Revert accidental commits to these files.
30922 2006-03-16 20:01:03 +0000 Michal Benes <michal.benes@xeris.cz>
30924 tests/Makefile.am: Don't try to build tests in tests/icles if we don't have X (#323852)
30925 Original commit message from CVS:
30926 Patch by: Michal Benes <michal dot benes at xeris dot cz>
30927 * tests/Makefile.am:
30928 Don't try to build tests in tests/icles if we
30929 don't have X (#323852)
30931 2006-03-16 13:08:01 +0000 Tim-Philipp Müller <tim@centricular.net>
30933 gst-libs/gst/tag/gstid3tag.c: Add TXXX frame identifiers for replaygain stuff as used by some taggers (see #323721).
30934 Original commit message from CVS:
30935 * gst-libs/gst/tag/gstid3tag.c:
30936 Add TXXX frame identifiers for replaygain stuff as used
30937 by some taggers (see #323721).
30939 2006-03-16 10:22:27 +0000 Jan Schmidt <thaytan@mad.scientist.com>
30941 gst/playback/gststreamselector.c: Preserve the existing buggy streamselector behaviour by performing a fallback buffe...
30942 Original commit message from CVS:
30943 * gst/playback/gststreamselector.c:
30944 (gst_stream_selector_set_property),
30945 (gst_stream_selector_bufferalloc):
30946 Preserve the existing buggy streamselector behaviour by performing
30947 a fallback buffer allocation when downstream isn't linked yet.
30948 This should really be fixed in playbin by blocking pads until it's
30950 Also, use gst_pad_alloc_buffer instead of
30951 gst_pad_alloc_buffer_and_set.
30953 2006-03-15 22:40:08 +0000 Tim-Philipp Müller <tim@centricular.net>
30955 gst-libs/gst/tag/gstid3tag.c: Don't crash on unknown ID3v2 TXXX frames.
30956 Original commit message from CVS:
30957 * gst-libs/gst/tag/gstid3tag.c:
30958 Don't crash on unknown ID3v2 TXXX frames.
30960 2006-03-15 17:59:05 +0000 Jan Schmidt <thaytan@mad.scientist.com>
30962 ext/alsa/gstalsasink.c: Chain up to the parent finalize method.
30963 Original commit message from CVS:
30964 * ext/alsa/gstalsasink.c: (gst_alsasink_finalise):
30965 Chain up to the parent finalize method.
30966 Add 32-bit sample size to the template caps.
30967 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
30968 (gst_riff_create_video_template_caps):
30969 Add the fourcc that the VMWare codec uses.
30970 * gst/playback/gststreamselector.c:
30971 (gst_stream_selector_set_property),
30972 (gst_stream_selector_bufferalloc),
30973 (gst_stream_selector_request_new_pad):
30974 For the active pad, forward buffer-alloc requests, otherwise
30975 return GST_FLOW_NOT_LINKED. This also prevents xvimagesink
30976 having to memcpy every frame when used by playbin.
30977 * gst/tcp/gstmultifdsink.c:
30978 (gst_multi_fd_sink_handle_client_write):
30979 Get negotiated caps from the sink pad, rather than the sink
30982 2006-03-15 17:11:34 +0000 Tommi Myöhänen <ext-tommi.myohanen@nokia.com>
30984 ext/gnomevfs/gstgnomevfssrc.c: Don't forget to set src->callbacks_pushed to FALSE again when popping them, otherwise ...
30985 Original commit message from CVS:
30986 Patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com>
30987 * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_pop_callbacks):
30988 Don't forget to set src->callbacks_pushed to FALSE again when
30989 popping them, otherwise re-activation in a different mode won't
30992 2006-03-15 11:30:29 +0000 Sebastien Moutte <sebastien@moutte.net>
30994 gst/ffmpegcolorspace/gstffmpegcodecmap.c: Replace __VA_ARGS__ caps creation macros with varargs functions. looks nice...
30995 Original commit message from CVS:
30996 Patch by: Sebastien Moutte <sebastien moutte net>
30997 * gst/ffmpegcolorspace/gstffmpegcodecmap.c: (gst_ff_vid_caps_new),
30998 (gst_ff_aud_caps_new), (gst_ffmpeg_pixfmt_to_caps),
30999 (gst_ffmpeg_smpfmt_to_caps):
31000 Replace __VA_ARGS__ caps creation macros with varargs functions.
31001 Makes things compile on MSVC (#320765), looks nicer, and we can
31002 tell the compiler to check for the NULL terminator.
31004 2006-03-14 15:13:04 +0000 Fabrizio Gennari <fabrizio.ge@tiscali.it>
31006 gst-libs/gst/riff/riff-media.c: Make sure the buffer we copy into is really always big enough, this time for real (#3...
31007 Original commit message from CVS:
31008 Patch by: Fabrizio Gennari <fabrizio dot ge at tiscali dot it>
31009 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
31010 Make sure the buffer we copy into is really always big
31011 enough, this time for real (#333488).
31013 2006-03-14 13:16:49 +0000 Tim-Philipp Müller <tim@centricular.net>
31015 gst-libs/gst/riff/riff-media.c: Add support for 24bpp DIB (#305279).
31016 Original commit message from CVS:
31017 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
31018 Add support for 24bpp DIB (#305279).
31020 2006-03-14 11:11:59 +0000 Wim Taymans <wim.taymans@gmail.com>
31022 gst/: Re-enable QoS after the release.
31023 Original commit message from CVS:
31024 * gst-libs/gst/video/gstvideofilter.c: (gst_video_filter_init):
31025 * gst-libs/gst/video/gstvideosink.c: (gst_video_sink_init):
31026 * gst/ffmpegcolorspace/gstffmpegcolorspace.c: (gst_ffmpegcsp_init):
31027 * gst/videoscale/gstvideoscale.c: (gst_video_scale_class_init),
31028 (gst_video_scale_init), (gst_video_scale_src_event):
31029 Re-enable QoS after the release.
31030 Rework videoscale to use the base class src_event handler.
31032 2006-03-14 09:51:01 +0000 Tim-Philipp Müller <tim@centricular.net>
31034 configure.ac: back to CVS.
31035 Original commit message from CVS:
31039 === release 0.10.5 ===
31041 2006-03-13 19:50:04 +0000 Thomas Vander Stichele <thomas@apestaart.org>
31047 * docs/plugins/inspect/plugin-adder.xml:
31048 * docs/plugins/inspect/plugin-alsa.xml:
31049 * docs/plugins/inspect/plugin-audioconvert.xml:
31050 * docs/plugins/inspect/plugin-audiorate.xml:
31051 * docs/plugins/inspect/plugin-audioresample.xml:
31052 * docs/plugins/inspect/plugin-audiotestsrc.xml:
31053 * docs/plugins/inspect/plugin-cdparanoia.xml:
31054 * docs/plugins/inspect/plugin-decodebin.xml:
31055 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
31056 * docs/plugins/inspect/plugin-gnomevfs.xml:
31057 * docs/plugins/inspect/plugin-libvisual.xml:
31058 * docs/plugins/inspect/plugin-ogg.xml:
31059 * docs/plugins/inspect/plugin-pango.xml:
31060 * docs/plugins/inspect/plugin-playbin.xml:
31061 * docs/plugins/inspect/plugin-subparse.xml:
31062 * docs/plugins/inspect/plugin-tcp.xml:
31063 * docs/plugins/inspect/plugin-theora.xml:
31064 * docs/plugins/inspect/plugin-typefindfunctions.xml:
31065 * docs/plugins/inspect/plugin-video4linux.xml:
31066 * docs/plugins/inspect/plugin-videorate.xml:
31067 * docs/plugins/inspect/plugin-videoscale.xml:
31068 * docs/plugins/inspect/plugin-videotestsrc.xml:
31069 * docs/plugins/inspect/plugin-volume.xml:
31070 * docs/plugins/inspect/plugin-vorbis.xml:
31071 * docs/plugins/inspect/plugin-ximagesink.xml:
31072 * docs/plugins/inspect/plugin-xvimagesink.xml:
31073 * win32/common/config.h:
31075 Original commit message from CVS:
31078 2006-03-13 17:28:23 +0000 Thomas Vander Stichele <thomas@apestaart.org>
31095 Original commit message from CVS:
31098 2006-03-13 11:17:19 +0000 Tim-Philipp Müller <tim@centricular.net>
31100 docs/plugins/Makefile.am: Part of previous cdparanoiasrc docs fixes, forgot to commit.
31101 Original commit message from CVS:
31102 * docs/plugins/Makefile.am:
31103 Part of previous cdparanoiasrc docs fixes, forgot to commit.
31105 2006-03-12 14:56:31 +0000 Tim-Philipp Müller <tim@centricular.net>
31107 docs/plugins/: Add cdparanoiasrc to docs.
31108 Original commit message from CVS:
31109 * docs/plugins/gst-plugins-base-plugins-docs.sgml:
31110 * docs/plugins/gst-plugins-base-plugins-sections.txt:
31111 * docs/plugins/gst-plugins-base-plugins.hierarchy:
31112 Add cdparanoiasrc to docs.
31113 * gst-libs/gst/cdda/gstcddabasesrc.c:
31114 More GstCddaBaseSrc docs.
31116 2006-03-12 13:47:22 +0000 Tim-Philipp Müller <tim@centricular.net>
31118 Add new API to libgsttag: gst_tag_from_id3_user_tag().
31119 Original commit message from CVS:
31120 * docs/libs/gst-plugins-base-libs-sections.txt:
31121 * gst-libs/gst/tag/gstid3tag.c: (gst_tag_from_id3_user_tag):
31122 * gst-libs/gst/tag/tag.h:
31123 Add new API to libgsttag: gst_tag_from_id3_user_tag().
31125 2006-03-11 19:47:16 +0000 Tim-Philipp Müller <tim@centricular.net>
31127 gst/typefind/gsttypefindfunctions.c: NULL-terminate array of mpeg4 video file extensions.
31128 Original commit message from CVS:
31129 * gst/typefind/gsttypefindfunctions.c: (plugin_init):
31130 NULL-terminate array of mpeg4 video file extensions.
31131 Fixes crash on PPC (#334226).
31133 2006-03-11 16:40:20 +0000 Tim-Philipp Müller <tim@centricular.net>
31135 ext/gnomevfs/gstgnomevfssrc.c: gnome_vfs_uri_is_local() alone is not a good indicator whether we can operate in pull-...
31136 Original commit message from CVS:
31137 * ext/gnomevfs/gstgnomevfssrc.c:
31138 (gst_gnome_vfs_src_check_get_range):
31139 gnome_vfs_uri_is_local() alone is not a good indicator
31140 whether we can operate in pull-mode with a specific URI,
31141 as it returns FALSE for file:// URIs that point to an
31142 NFS-mounted path. Be more conservative here: whitelist
31143 local files, blacklist http URIs and use the old
31144 mechanism for anything else (fixes #334216).
31146 2006-03-10 19:15:34 +0000 Thomas Vander Stichele <thomas@apestaart.org>
31148 configure.ac: back to trunk
31149 Original commit message from CVS:
31153 === release 0.10.4 ===
31155 2006-03-10 19:05:13 +0000 Thomas Vander Stichele <thomas@apestaart.org>
31161 * docs/plugins/gst-plugins-base-plugins.args:
31162 * docs/plugins/inspect/plugin-adder.xml:
31163 * docs/plugins/inspect/plugin-alsa.xml:
31164 * docs/plugins/inspect/plugin-audioconvert.xml:
31165 * docs/plugins/inspect/plugin-audiorate.xml:
31166 * docs/plugins/inspect/plugin-audioresample.xml:
31167 * docs/plugins/inspect/plugin-audiotestsrc.xml:
31168 * docs/plugins/inspect/plugin-cdparanoia.xml:
31169 * docs/plugins/inspect/plugin-decodebin.xml:
31170 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
31171 * docs/plugins/inspect/plugin-gnomevfs.xml:
31172 * docs/plugins/inspect/plugin-libvisual.xml:
31173 * docs/plugins/inspect/plugin-ogg.xml:
31174 * docs/plugins/inspect/plugin-pango.xml:
31175 * docs/plugins/inspect/plugin-playbin.xml:
31176 * docs/plugins/inspect/plugin-subparse.xml:
31177 * docs/plugins/inspect/plugin-tcp.xml:
31178 * docs/plugins/inspect/plugin-theora.xml:
31179 * docs/plugins/inspect/plugin-typefindfunctions.xml:
31180 * docs/plugins/inspect/plugin-video4linux.xml:
31181 * docs/plugins/inspect/plugin-videorate.xml:
31182 * docs/plugins/inspect/plugin-videoscale.xml:
31183 * docs/plugins/inspect/plugin-videotestsrc.xml:
31184 * docs/plugins/inspect/plugin-volume.xml:
31185 * docs/plugins/inspect/plugin-vorbis.xml:
31186 * docs/plugins/inspect/plugin-ximagesink.xml:
31187 * docs/plugins/inspect/plugin-xvimagesink.xml:
31189 * win32/common/config.h:
31191 Original commit message from CVS:
31194 2006-03-10 12:37:53 +0000 Jan Schmidt <thaytan@mad.scientist.com>
31196 gst-libs/gst/video/gstvideosink.c: Disable max-lateness by setting it to -1 for now, so that we can bed QoS stuff in ...
31197 Original commit message from CVS:
31198 * gst-libs/gst/video/gstvideosink.c: (gst_video_sink_init):
31199 Disable max-lateness by setting it to -1 for now, so that
31200 we can bed QoS stuff in thoroughly between now and the next
31203 2006-03-10 11:09:23 +0000 Fabrizio <fabrizio.ge@tiscali.it>
31205 gst-libs/gst/riff/riff-media.c: Make sure we don't read beyond the palette buffer in case of
31206 Original commit message from CVS:
31207 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
31208 Make sure we don't read beyond the palette buffer in case of
31209 broken or manipulated files (#333488, patch by: Fabrizio
31212 2006-03-10 10:44:02 +0000 Edward Hervey <bilboed@bilboed.com>
31214 gst/typefind/gsttypefindfunctions.c: Fix for variable not initialized.
31215 Original commit message from CVS:
31216 * gst/typefind/gsttypefindfunctions.c: (mp3_type_find_at_offset):
31217 Fix for variable not initialized.
31219 2006-03-09 19:02:32 +0000 Thomas Vander Stichele <thomas@apestaart.org>
31222 * docs/libs/tmpl/gstringbuffer.sgml:
31237 * win32/common/config.h:
31239 Original commit message from CVS:
31242 2006-03-09 17:58:00 +0000 Wim Taymans <wim.taymans@gmail.com>
31244 ext/libvisual/visual.c: Small cleanups.
31245 Original commit message from CVS:
31246 * ext/libvisual/visual.c: (gst_visual_get_type),
31247 (gst_visual_src_setcaps), (gst_vis_src_negotiate),
31248 (gst_visual_chain):
31250 * ext/theora/gsttheoradec.h:
31251 * ext/theora/theoradec.c: (gst_theora_dec_init),
31252 (gst_theora_dec_reset), (_theora_granule_time),
31253 (theora_dec_src_convert), (theora_dec_sink_convert),
31254 (theora_dec_src_query), (theora_dec_src_event),
31255 (theora_dec_sink_event), (theora_handle_comment_packet),
31256 (theora_handle_header_packet), (theora_dec_push),
31257 (theora_handle_data_packet), (theora_dec_chain),
31258 (theora_dec_change_state):
31261 2006-03-09 17:50:59 +0000 Wim Taymans <wim.taymans@gmail.com>
31263 ext/gnomevfs/gstgnomevfssrc.c: Some cleanups.
31264 Original commit message from CVS:
31265 * ext/gnomevfs/gstgnomevfssrc.c: (audiocast_init),
31266 (audiocast_register_listener), (gst_gnome_vfs_src_start):
31269 2006-03-09 17:45:39 +0000 Wim Taymans <wim.taymans@gmail.com>
31271 ext/ogg/gstoggdemux.c: Don't try to activate NULL chains.
31272 Original commit message from CVS:
31273 * ext/ogg/gstoggdemux.c: (gst_ogg_demux_activate_chain):
31274 Don't try to activate NULL chains.
31276 2006-03-09 16:30:41 +0000 Tim-Philipp Müller <tim@centricular.net>
31278 gst/typefind/gsttypefindfunctions.c: Fix invalid memory access to region before peek'd data (#332964).
31279 Original commit message from CVS:
31280 * gst/typefind/gsttypefindfunctions.c: (mp3_type_find_at_offset):
31281 Fix invalid memory access to region before peek'd data (#332964).
31283 2006-03-09 15:05:03 +0000 Christophe Fergeau <teuf@gnome.org>
31286 Original commit message from CVS:
31287 * ext/pango/gsttextoverlay.c: (gst_text_overlay_init):
31288 * ext/pango/gsttextrender.c: (gst_text_render_init):
31289 * gst/adder/gstadder.c: (gst_adder_init):
31290 Don't leak padtemplates, patch by Christophe Fergeau,
31293 2006-03-09 12:56:35 +0000 Tim-Philipp Müller <tim@centricular.net>
31295 gst/subparse/gstsubparse.c: Fix invalid memory access: make sure string passed to regexec() is NUL-termianted.
31296 Original commit message from CVS:
31297 * gst/subparse/gstsubparse.c: (gst_subparse_type_find):
31298 Fix invalid memory access: make sure string passed to
31299 regexec() is NUL-termianted.
31301 2006-03-09 12:37:59 +0000 Tim-Philipp Müller <tim@centricular.net>
31303 gst/typefind/gsttypefindfunctions.c: Refactor mpeg/audio typefinding to make it more maintainable and easier to fine-...
31304 Original commit message from CVS:
31305 * gst/typefind/gsttypefindfunctions.c: (mp3_type_find_at_offset),
31307 Refactor mpeg/audio typefinding to make it more maintainable
31308 and easier to fine-tune. Make probing into middle of the file
31309 work properly (fixes #333900, also see #152688).
31311 2006-03-09 11:10:03 +0000 Tim-Philipp Müller <tim@centricular.net>
31313 gst/typefind/gsttypefindfunctions.c: Remove part from previous commit that was bogus: g_utf8_validate() does in fact ...
31314 Original commit message from CVS:
31315 * gst/typefind/gsttypefindfunctions.c:
31316 (utf8_type_find_have_valid_utf8_at_offset):
31317 Remove part from previous commit that was bogus:
31318 g_utf8_validate() does in fact not accept embedded
31319 zeroes, so we don't need to check for those (thanks
31320 to Mike for the hint).
31322 2006-03-08 17:11:29 +0000 Tim-Philipp Müller <tim@centricular.net>
31324 gst/typefind/gsttypefindfunctions.c: Make plain/text typefinder more conservative: firstly, check for embedded zeroes...
31325 Original commit message from CVS:
31326 * gst/typefind/gsttypefindfunctions.c:
31327 (utf8_type_find_count_embedded_zeroes),
31328 (utf8_type_find_have_valid_utf8_at_offset), (utf8_type_find):
31329 Make plain/text typefinder more conservative: firstly, check
31330 for embedded zeroes, which are perfectly valid UTF-8 characters,
31331 but also a fairly good sign that something is not a plain text
31332 file; secondly, probe into the middle of the file if possible.
31333 If we can't probe into the middle, limit the probability value
31334 to be returned to TYPE_FIND_POSSIBLE (see #333900).
31336 2006-03-08 11:34:45 +0000 Michael Smith <msmith@xiph.org>
31338 gst/typefind/gsttypefindfunctions.c: Make typefind function name for mpeg4 video unique.
31339 Original commit message from CVS:
31340 * gst/typefind/gsttypefindfunctions.c: (plugin_init):
31341 Make typefind function name for mpeg4 video unique.
31343 2006-03-08 09:53:31 +0000 Wim Taymans <wim.taymans@gmail.com>
31345 ext/libvisual/visual.c: Cleanups, post nice errors.
31346 Original commit message from CVS:
31347 * ext/libvisual/visual.c: (gst_visual_init),
31348 (gst_visual_clear_actors), (gst_visual_dispose),
31349 (gst_visual_reset), (gst_visual_src_setcaps),
31350 (gst_visual_sink_setcaps), (gst_vis_src_negotiate),
31351 (gst_visual_sink_event), (gst_visual_src_event), (get_buffer),
31352 (gst_visual_chain), (gst_visual_change_state):
31353 Cleanups, post nice errors.
31354 Handle sink and src events.
31355 Implement simple QoS.
31356 * gst-libs/gst/video/gstvideosink.c: (gst_video_sink_init):
31357 Use new basesink methods to configure max-lateness.
31359 * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
31360 (gst_ffmpegcsp_transform_caps), (gst_ffmpegcsp_set_caps):
31361 Debug statement cleanups.
31362 * gst/volume/gstvolume.c: (gst_volume_class_init):
31365 2006-03-08 09:50:23 +0000 Tim-Philipp Müller <tim@centricular.net>
31367 ext/pango/gsttextoverlay.c: Revert API/ABI break from March 1. Keep 'halign' and 'valign' as string type properties, ...
31368 Original commit message from CVS:
31369 * ext/pango/gsttextoverlay.c: (gst_text_overlay_class_init),
31370 (gst_text_overlay_init), (gst_text_overlay_set_property),
31371 (gst_text_overlay_get_property):
31372 Revert API/ABI break from March 1. Keep 'halign' and 'valign'
31373 as string type properties, but mark them deprecated. Add
31374 'halignment' and 'valignment' properties that use enums
31375 instead of strings.
31377 2006-03-08 09:37:12 +0000 Fabrizio <fabrizio.ge@tiscali.it>
31379 gst-libs/gst/riff/riff-media.c: Allow palettes with less than 256 colours in AVI files
31380 Original commit message from CVS:
31381 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
31382 Allow palettes with less than 256 colours in AVI files
31383 (#333488, patch by: Fabrizio Gennari).
31385 2006-03-07 21:56:09 +0000 Julien Moutte <julien@moutte.net>
31387 ext/pango/gsttextoverlay.c: Fix wrong EOS handling on text pad. We were releasing the queued text buffer when we shou...
31388 Original commit message from CVS:
31389 2006-03-07 Julien MOUTTE <julien@moutte.net>
31390 * ext/pango/gsttextoverlay.c: (gst_text_overlay_text_event),
31391 (gst_text_overlay_video_event): Fix wrong EOS handling on text
31392 pad. We were releasing the queued text buffer when we should keep
31393 it until video pad gets EOS or discard the text buffer because it's
31394 too old. That was eating the last subtitle buffer. Add some more
31397 2006-03-07 17:28:36 +0000 Tim-Philipp Müller <tim@centricular.net>
31399 ext/pango/gsttextoverlay.c: Fix invalid memory access (we can't access a buffer after it's been pushed downstream wit...
31400 Original commit message from CVS:
31401 * ext/pango/gsttextoverlay.c: (gst_text_overlay_render_text),
31402 (gst_text_overlay_video_chain):
31403 Fix invalid memory access (we can't access a buffer after it's been
31404 pushed downstream without taking a reference); fix memory leak (if
31405 there's no text to render, bail out before allocating stuff).
31407 2006-03-07 15:08:15 +0000 Tim-Philipp Müller <tim@centricular.net>
31409 ext/pango/gsttextoverlay.*: If input is plain text, escape it before passing it to pango_layout_set_markup().
31410 Original commit message from CVS:
31411 * ext/pango/gsttextoverlay.c: (gst_text_overlay_init),
31412 (gst_text_overlay_setcaps_txt), (gst_text_overlay_video_chain):
31413 * ext/pango/gsttextoverlay.h:
31414 If input is plain text, escape it before passing it to
31415 pango_layout_set_markup().
31417 2006-03-07 13:01:21 +0000 Tim-Philipp Müller <tim@centricular.net>
31419 gst-libs/gst/audio/gstaudiofilter.c: Don't ignore flow return from gst_pad_push().
31420 Original commit message from CVS:
31421 * gst-libs/gst/audio/gstaudiofilter.c: (gst_audio_filter_chain):
31422 Don't ignore flow return from gst_pad_push().
31424 2006-03-07 12:49:03 +0000 Christophe Fergeau <teuf@gnome.org>
31426 Don't leak references returned by gst_pad_get_parent()
31427 Original commit message from CVS:
31428 * ext/libvisual/visual.c: (gst_visual_getcaps),
31429 (gst_visual_src_setcaps), (gst_visual_sink_setcaps):
31430 * ext/ogg/gstoggmux.c: (gst_ogg_mux_sinkconnect):
31431 * ext/vorbis/vorbisenc.c: (gst_vorbisenc_convert_src),
31432 (gst_vorbisenc_convert_sink):
31433 * gst-libs/gst/audio/audio.c: (gst_audio_frame_byte_size),
31434 (gst_audio_duration_from_pad_buffer):
31435 * gst-libs/gst/audio/gstaudiofilter.c: (gst_audio_filter_link),
31436 (gst_audio_filter_chain):
31437 * gst-libs/gst/rtp/gstbasertpdepayload.c:
31438 (gst_base_rtp_depayload_setcaps):
31439 * gst-libs/gst/video/video.c: (gst_video_frame_rate),
31440 (gst_video_get_size):
31441 * gst/audiorate/gstaudiorate.c: (gst_audio_rate_setcaps):
31442 Don't leak references returned by gst_pad_get_parent()
31443 (#333663, based on patch by: Christophe Fergeau).
31445 2006-03-06 20:52:25 +0000 Stefan Kost <ensonic@users.sourceforge.net>
31447 ext/gnomevfs/gstgnomevfssink.c: change location param details
31448 Original commit message from CVS:
31449 * ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_class_init):
31450 change location param details
31451 * gst/volume/gstvolume.c: (plugin_init):
31452 correct plugin description
31454 2006-03-06 20:07:55 +0000 Tim-Philipp Müller <tim@centricular.net>
31456 ext/gnomevfs/gstgnomevfssrc.c: Override GstBaseSrc::check_get_range() in order to avoid opening the resource just to ...
31457 Original commit message from CVS:
31458 * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_class_init),
31459 (gst_gnome_vfs_src_check_get_range):
31460 Override GstBaseSrc::check_get_range() in order to avoid opening
31461 the resource just to check whether we can operate in pull-mode or
31462 not - we can predict that pretty well from the URI alone. Should
31463 fix problems with last.fm (#331690). (Requires latest core CVS).
31465 2006-03-06 16:18:51 +0000 Wim Taymans <wim.taymans@gmail.com>
31467 gst-libs/gst/video/gstvideosink.c: Throw away frames that are later than 20 ms.
31468 Original commit message from CVS:
31469 * gst-libs/gst/video/gstvideosink.c: (gst_video_sink_init),
31470 (gst_video_sink_class_init):
31471 Throw away frames that are later than 20 ms.
31473 2006-03-06 14:14:47 +0000 Fabrizio <fabrizio.ge@tiscali.it>
31475 gst-libs/gst/riff/riff-media.c:
31476 Original commit message from CVS:
31477 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
31478 Set depth on WMA caps (#333545, patch by: Fabrizio Gennari).
31480 2006-03-05 23:39:50 +0000 Thomas Vander Stichele <thomas@apestaart.org>
31482 ext/ogg/gstoggmux.c: put Theora BOS pages before others. This hardcodes the Ogg/Theora I profile, but hey.
31483 Original commit message from CVS:
31484 * ext/ogg/gstoggmux.c: (gst_ogg_mux_buffer_from_page),
31485 (gst_ogg_mux_send_headers), (gst_ogg_mux_collected):
31486 put Theora BOS pages before others. This hardcodes
31487 the Ogg/Theora I profile, but hey.
31489 2006-03-05 23:06:22 +0000 Thomas Vander Stichele <thomas@apestaart.org>
31491 * ext/ogg/gstoggmux.c:
31492 changed more than 5 lines
31493 Original commit message from CVS:
31494 changed more than 5 lines
31496 2006-03-05 22:57:58 +0000 Thomas Vander Stichele <thomas@apestaart.org>
31498 ogg muxing of vorbis and theora now has pages ordered correctly again, even with delays.
31499 Original commit message from CVS:
31500 ogg muxing of vorbis and theora now has pages ordered correctly again,
31503 updated with some examples
31504 * ext/theora/theoraenc.c: (granulepos_to_timestamp),
31505 (granulepos_add), (theora_buffer_from_packet):
31506 * ext/vorbis/vorbisenc.c: (granulepos_to_timestamp_offset),
31507 (granulepos_to_timestamp), (gst_vorbisenc_buffer_from_packet),
31508 (gst_vorbisenc_chain):
31509 implement strategy from ext/ogg/README
31510 * ext/ogg/gstoggmux.c: (gst_ogg_mux_buffer_from_page),
31511 (gst_ogg_mux_push_buffer), (gst_ogg_mux_dequeue_page),
31512 (gst_ogg_mux_pad_queue_page), (gst_ogg_mux_compare_pads),
31513 (gst_ogg_mux_queue_pads), (gst_ogg_mux_collected):
31514 Fix muxer so that oggz-validate is happy with all streams;
31515 except for no eos mark, and the BOS page ordering
31516 * tests/check/pipelines/theoraenc.c: (check_buffer_is_header),
31517 (check_buffer_granulepos):
31518 * tests/check/pipelines/vorbisenc.c: (check_buffer_granulepos):
31519 update tests to check for OFFSET being set as requested
31520 fixed type of granulepos, it's not a ClockTime
31522 2006-03-05 21:34:23 +0000 Julien Moutte <julien@moutte.net>
31524 sys/xvimage/xvimagesink.c: Check that the xvimage we are creating has a correct size before returning it. (#3...
31525 Original commit message from CVS:
31526 2006-03-05 Julien MOUTTE <julien@moutte.net>
31527 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_new),
31528 (gst_xvimagesink_show_frame), (gst_xvimagesink_buffer_alloc):
31529 Check that the xvimage we are creating has a correct size before returning it. (#314897)
31531 2006-03-05 13:44:05 +0000 Tim-Philipp Müller <tim@centricular.net>
31533 gst/typefind/gsttypefindfunctions.c: Give id3 and ape tag typefinders a rank slightly higher than PRIMARY to ensure t...
31534 Original commit message from CVS:
31535 * gst/typefind/gsttypefindfunctions.c: (plugin_init):
31536 Give id3 and ape tag typefinders a rank slightly higher
31537 than PRIMARY to ensure they're always run before any of
31538 the other typefinders (in particular wav and mp3) (#324186).
31540 2006-03-05 13:08:37 +0000 Tim-Philipp Müller <tim@centricular.net>
31542 gst-libs/gst/riff/riff-media.c: Add support for '3IVD' fourcc (#333403).
31543 Original commit message from CVS:
31544 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
31545 Add support for '3IVD' fourcc (#333403).
31547 2006-03-04 14:35:10 +0000 Tim-Philipp Müller <tim@centricular.net>
31549 configure.ac: Bump requirements to GStreamer CVS for the new error enum.
31550 Original commit message from CVS:
31552 Bump requirements to GStreamer CVS for the new error enum.
31553 * ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_render):
31554 Emit new GST_RESOURCE_ERROR_NO_SPACE_LEFT when there's no
31555 space left on the device (fixes #333352).
31557 2006-03-03 23:53:50 +0000 Sébastien Moutte <sebastien@moutte.net>
31559 win32/vs6: add a project file for libgstvolume update the workspace
31560 Original commit message from CVS:
31562 add a project file for libgstvolume
31563 update the workspace
31565 2006-03-03 15:26:57 +0000 Thomas Vander Stichele <thomas@apestaart.org>
31568 * ext/ogg/gstoggmux.c:
31570 Original commit message from CVS:
31573 2006-03-03 15:22:02 +0000 Thomas Vander Stichele <thomas@apestaart.org>
31575 Fix for http://bugzilla.gnome.org/show_bug.cgi?id=333254
31576 Original commit message from CVS:
31577 2006-03-03 Thomas Vander Stichele <thomas at apestaart dot org>
31578 * ext/theora/theoraenc.c: (theora_set_header_on_caps):
31579 * tests/check/pipelines/theoraenc.c: (check_buffer_is_header),
31581 Fix for http://bugzilla.gnome.org/show_bug.cgi?id=333254
31582 Set IN_CAPS on header buffers
31584 2006-03-02 18:23:55 +0000 Wim Taymans <wim.taymans@gmail.com>
31586 docs/plugins/: Add audioresample to docs.
31587 Original commit message from CVS:
31588 * docs/plugins/Makefile.am:
31589 * docs/plugins/gst-plugins-base-plugins-docs.sgml:
31590 * docs/plugins/gst-plugins-base-plugins-sections.txt:
31591 Add audioresample to docs.
31592 * gst/audioconvert/gstaudioconvert.c:
31594 * gst/audioresample/gstaudioresample.c:
31595 (gst_audioresample_base_init), (gst_audioresample_class_init),
31596 (gst_audioresample_init), (gst_audioresample_dispose),
31597 (audioresample_get_unit_size), (audioresample_transform_caps),
31598 (resample_set_state_from_caps), (audioresample_transform_size),
31599 (audioresample_set_caps), (audioresample_event),
31600 (audioresample_do_output), (audioresample_transform),
31601 (audioresample_pushthrough), (gst_audioresample_set_property),
31602 (gst_audioresample_get_property), (plugin_init):
31603 * gst/audioresample/gstaudioresample.h:
31605 Small code cleanups.
31607 2006-03-02 18:12:33 +0000 Thomas Vander Stichele <thomas@apestaart.org>
31609 * gst/videorate/Makefile.am:
31611 Original commit message from CVS:
31614 2006-03-02 17:48:40 +0000 Thomas Vander Stichele <thomas@apestaart.org>
31616 * ext/ogg/gstoggmux.c:
31617 debug using the actual GstPad, that allows us to see the serialno in the padname
31618 Original commit message from CVS:
31619 debug using the actual GstPad, that allows us to see the serialno in the padname
31621 2006-03-02 17:46:36 +0000 Wim Taymans <wim.taymans@gmail.com>
31623 docs/plugins/: Added videoscale to docs.
31624 Original commit message from CVS:
31625 * docs/plugins/Makefile.am:
31626 * docs/plugins/gst-plugins-base-plugins-docs.sgml:
31627 * docs/plugins/gst-plugins-base-plugins-sections.txt:
31628 Added videoscale to docs.
31629 * gst/videorate/gstvideorate.c: (gst_video_rate_flush_prev),
31630 (gst_video_rate_swap_prev), (gst_video_rate_event),
31631 (gst_video_rate_chain):
31633 * gst/videoscale/gstvideoscale.c: (gst_video_scale_class_init),
31634 (gst_video_scale_init), (gst_video_scale_prepare_size),
31635 (gst_video_scale_set_caps), (gst_video_scale_get_unit_size),
31636 (gst_video_scale_fixate_caps), (gst_video_scale_transform):
31637 * gst/videoscale/gstvideoscale.h:
31638 Added docs, examples.
31639 Some code cleanups.
31640 Post errors instead of g_warning.
31642 2006-03-02 17:30:57 +0000 Thomas Vander Stichele <thomas@apestaart.org>
31644 * ext/ogg/gstoggmux.c:
31645 clean up debug messages
31646 Original commit message from CVS:
31647 clean up debug messages
31649 2006-03-02 17:15:38 +0000 Thomas Vander Stichele <thomas@apestaart.org>
31651 * ext/ogg/gstoggmux.c:
31652 extra debugging from older version, makes it easier to compare
31653 Original commit message from CVS:
31654 extra debugging from older version, makes it easier to compare
31656 2006-03-02 17:04:55 +0000 Thomas Vander Stichele <thomas@apestaart.org>
31658 * ext/ogg/gstoggmux.c:
31659 some space cleanup and debug fixes
31660 Original commit message from CVS:
31661 some space cleanup and debug fixes
31663 2006-03-02 16:47:34 +0000 Wim Taymans <wim.taymans@gmail.com>
31665 docs/: Added some more docs to libs and plugins.
31666 Original commit message from CVS:
31667 * docs/libs/gst-plugins-base-libs-docs.sgml:
31668 * docs/libs/gst-plugins-base-libs-sections.txt:
31669 * docs/libs/gst-plugins-base-libs.types:
31670 * docs/plugins/Makefile.am:
31671 * docs/plugins/gst-plugins-base-plugins-docs.sgml:
31672 * docs/plugins/gst-plugins-base-plugins-sections.txt:
31673 Added some more docs to libs and plugins.
31674 * gst-libs/gst/audio/gstringbuffer.c:
31675 (gst_ring_buffer_prepare_read), (gst_ring_buffer_clear):
31676 * gst-libs/gst/audio/gstringbuffer.h:
31677 Document ringbuffer some more.
31678 * gst/videorate/gstvideorate.c: (gst_video_rate_class_init),
31679 (gst_video_rate_setcaps), (gst_video_rate_reset),
31680 (gst_video_rate_init), (gst_video_rate_flush_prev),
31681 (gst_video_rate_swap_prev), (gst_video_rate_event),
31682 (gst_video_rate_chain), (gst_video_rate_change_state):
31683 * gst/videorate/gstvideorate.h:
31684 Fix videorate to use segments.
31685 Make it work with 0/1 framerates (closes #331903)
31686 Handle EOS correctly.
31689 2006-03-02 13:13:00 +0000 Tim-Philipp Müller <tim@centricular.net>
31691 ext/ogg/gstogmparse.c: In state change function, first chain up to parent class, then handle downwards state change s...
31692 Original commit message from CVS:
31693 * ext/ogg/gstogmparse.c: (gst_ogm_parse_class_init),
31694 (gst_ogm_audio_parse_init), (gst_ogm_video_parse_init),
31695 (gst_ogm_text_parse_init), (gst_ogm_parse_change_state):
31696 In state change function, first chain up to parent class,
31697 then handle downwards state change stuff. Remove some
31698 commented out cruft from 0.8 code.
31700 2006-03-02 12:35:59 +0000 Tim-Philipp Müller <tim@centricular.net>
31702 ext/ogg/gstogmparse.c: Don't remove/re-add source pad if the new caps are the same as the old caps anyway (#333042). ...
31703 Original commit message from CVS:
31704 * ext/ogg/gstogmparse.c: (gst_ogm_audio_parse_init),
31705 (gst_ogm_video_parse_init), (gst_ogm_text_parse_init),
31706 (gst_ogm_parse_sink_convert), (gst_ogm_parse_sink_query),
31707 (gst_ogm_parse_chain):
31708 Don't remove/re-add source pad if the new caps are the same as
31709 the old caps anyway (#333042). When removing source pad, don't
31710 unref it afterwards - we didn't ref it when adding. Sprinkle some
31711 GST_DEBUG_FUNCPTR goodness here and there. Don't leak references
31712 after using gst_pad_get_parent(). Return downstream flow return
31713 value in chain function.
31715 2006-03-02 11:28:23 +0000 Wim Taymans <wim.taymans@gmail.com>
31717 docs/plugins/: Fix hierarchy, added some more elements to the docs.
31718 Original commit message from CVS:
31719 * docs/plugins/gst-plugins-base-plugins-docs.sgml:
31720 * docs/plugins/gst-plugins-base-plugins.args:
31721 * docs/plugins/gst-plugins-base-plugins.hierarchy:
31722 * docs/plugins/gst-plugins-base-plugins.interfaces:
31723 * docs/plugins/gst-plugins-base-plugins.signals:
31724 Fix hierarchy, added some more elements to the docs.
31725 * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
31726 (gst_ffmpegcsp_get_type):
31727 * gst/ffmpegcolorspace/gstffmpegcolorspace.h:
31728 Fix docs for ffmpegcolorspace.
31730 2006-03-01 19:24:44 +0000 Tim-Philipp Müller <tim@centricular.net>
31732 gst/typefind/gsttypefindfunctions.c: Some typefinding fine-tuning:
31733 Original commit message from CVS:
31734 * gst/typefind/gsttypefindfunctions.c: (id3_type_find),
31735 (apetag_type_find), (ape_type_find), (plugin_init):
31736 Some typefinding fine-tuning:
31737 - rank ID3/APE tags in order of preference via probabilities, so that
31738 ID3v2 > APEv2 > APEv1 > ID3v1.
31739 - three or four bytes don't really justify MAXIMUM probability,
31740 change those to 'very likely' (musepack and monkeysaudio).
31742 2006-03-01 18:25:18 +0000 Wim Taymans <wim.taymans@gmail.com>
31745 Original commit message from CVS:
31746 * docs/plugins/Makefile.am:
31747 * docs/plugins/gst-plugins-base-plugins-docs.sgml:
31748 * docs/plugins/gst-plugins-base-plugins-sections.txt:
31749 * ext/alsa/gstalsamixer.c:
31750 * ext/alsa/gstalsamixer.h:
31751 * ext/alsa/gstalsamixerelement.c:
31752 (gst_alsa_mixer_element_class_init), (gst_alsa_mixer_element_init):
31753 * ext/alsa/gstalsamixerelement.h:
31754 * ext/alsa/gstalsasink.c:
31755 * ext/alsa/gstalsasink.h:
31756 * ext/alsa/gstalsasrc.c: (gst_alsasrc_class_init),
31757 (gst_alsasrc_init):
31758 * ext/alsa/gstalsasrc.h:
31760 Small code cleanups.
31762 2006-03-01 17:52:45 +0000 Wim Taymans <wim.taymans@gmail.com>
31764 ext/theora/Makefile.am: Dist new header too,
31765 Original commit message from CVS:
31766 * ext/theora/Makefile.am:
31767 Dist new header too,
31769 2006-03-01 17:39:28 +0000 Wim Taymans <wim.taymans@gmail.com>
31771 Fix some more docs.
31772 Original commit message from CVS:
31773 * docs/plugins/Makefile.am:
31774 * docs/plugins/gst-plugins-base-plugins-docs.sgml:
31775 * docs/plugins/gst-plugins-base-plugins-sections.txt:
31776 * ext/gnomevfs/gstgnomevfssink.h:
31777 * ext/gnomevfs/gstgnomevfssrc.h:
31778 * ext/vorbis/vorbisdec.c: (vorbis_handle_data_packet):
31779 * ext/vorbis/vorbisdec.h:
31780 * ext/vorbis/vorbisenc.c: (gst_vorbisenc_convert_sink):
31781 * ext/vorbis/vorbisenc.h:
31782 * ext/vorbis/vorbisparse.c: (vorbis_parse_set_header_on_caps),
31783 (vorbis_parse_chain), (vorbis_parse_change_state):
31784 * ext/vorbis/vorbisparse.h:
31785 * gst/audioconvert/gstaudioconvert.h:
31786 * gst/tcp/gsttcpserversink.h:
31787 * gst/videotestsrc/gstvideotestsrc.c:
31788 * gst/videotestsrc/gstvideotestsrc.h:
31789 * gst/volume/gstvolume.c:
31790 * gst/volume/gstvolume.h:
31791 Fix some more docs.
31792 Added docs for vorbisdec and vorbisparse.
31795 2006-03-01 16:24:37 +0000 Wim Taymans <wim.taymans@gmail.com>
31797 Updated/added documentation.
31798 Original commit message from CVS:
31799 * docs/plugins/Makefile.am:
31800 * docs/plugins/gst-plugins-base-plugins-docs.sgml:
31801 * docs/plugins/gst-plugins-base-plugins-sections.txt:
31802 * ext/pango/gstclockoverlay.h:
31803 * ext/pango/gsttextoverlay.h:
31804 * ext/pango/gsttextrender.h:
31805 * ext/pango/gsttimeoverlay.h:
31806 * ext/theora/gsttheoradec.h:
31807 * ext/theora/gsttheoraenc.h:
31808 * ext/theora/theoradec.c:
31809 * ext/theora/theoraenc.c:
31810 * gst/audioconvert/gstaudioconvert.h:
31811 * gst/audiotestsrc/gstaudiotestsrc.h:
31812 * gst/ffmpegcolorspace/gstffmpegcolorspace.h:
31813 * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init):
31814 * gst/tcp/gstmultifdsink.h:
31815 Updated/added documentation.
31816 * ext/pango/gsttextoverlay.c: (gst_text_overlay_valign_get_type),
31817 (gst_text_overlay_halign_get_type),
31818 (gst_text_overlay_wrap_mode_get_type),
31819 (gst_text_overlay_base_init), (gst_text_overlay_class_init),
31820 (gst_text_overlay_init), (gst_text_overlay_set_property),
31821 (gst_text_overlay_get_property):
31822 Fix up properties to be enums instead of string to make bindings,
31823 introspection and automatic GUI creation possible.
31824 Add getters for the properties.
31826 2006-02-28 21:21:07 +0000 Sébastien Moutte <sebastien@moutte.net>
31828 gst/audiotestsrc/gstaudiotestsrc.c: added defines of M_PI and M_PI_2
31829 Original commit message from CVS:
31830 * gst/audiotestsrc/gstaudiotestsrc.c:
31831 added defines of M_PI and M_PI_2
31832 * gst/ffmpegcolorspace/avcodec.h:
31833 removed #include "stdint.h" for win32 as _stdint.h is
31834 autogenerated to win32/common
31835 * win32/common/libgstaudio.def:
31836 * win32/common/libgsttag.def:
31839 some project files bugs corrected
31841 project files are reset to the default vs7 configuration
31842 (they link to msvcr71.dll using default optimizations)
31844 2006-02-28 19:08:12 +0000 Wim Taymans <wim.taymans@gmail.com>
31846 ext/gnomevfs/gstgnomevfssink.c: Fix some docs.
31847 Original commit message from CVS:
31848 * ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_class_init):
31851 2006-02-28 13:52:04 +0000 Edward Hervey <bilboed@bilboed.com>
31853 ext/alsa/gstalsasrc.c: Set proper class on the ElementDetails:
31854 Original commit message from CVS:
31855 * ext/alsa/gstalsasrc.c:
31856 Set proper class on the ElementDetails:
31857 Source/Audio instead of Src/Audio
31859 2006-02-28 12:19:11 +0000 Edward Hervey <bilboed@bilboed.com>
31861 gst/videoscale/vs_scanline.c: Revert optimization in videoscale. It should go in liboil and have an appropriate liboi...
31862 Original commit message from CVS:
31863 * gst/videoscale/vs_scanline.c:
31864 (vs_scanline_resample_nearest_RGBA):
31865 Revert optimization in videoscale. It should go in liboil and have
31866 an appropriate liboil function.
31868 2006-02-28 11:06:24 +0000 Wim Taymans <wim.taymans@gmail.com>
31870 gst-libs/gst/audio/gstbaseaudiosink.c: Don't try to provide a clock in the NULL state.
31871 Original commit message from CVS:
31872 * gst-libs/gst/audio/gstbaseaudiosink.c:
31873 (gst_base_audio_sink_provide_clock):
31874 Don't try to provide a clock in the NULL state.
31876 2006-02-28 11:04:47 +0000 Wim Taymans <wim.taymans@gmail.com>
31878 ext/ogg/gstoggdemux.c: Use GstSegment infrastructure to remove duplicated code and handle more seek cases correctly.
31879 Original commit message from CVS:
31880 * ext/ogg/gstoggdemux.c: (gst_ogg_demux_receive_event),
31881 (gst_ogg_pad_event), (gst_ogg_pad_internal_chain),
31882 (gst_ogg_demux_chain_peer), (gst_ogg_pad_submit_packet),
31883 (gst_ogg_demux_deactivate_current_chain),
31884 (gst_ogg_demux_activate_chain), (gst_ogg_demux_do_seek),
31885 (gst_ogg_demux_perform_seek), (gst_ogg_demux_collect_info),
31886 (gst_ogg_demux_find_chains), (gst_ogg_demux_chain),
31887 (gst_ogg_demux_loop), (gst_ogg_demux_change_state):
31888 Use GstSegment infrastructure to remove duplicated code
31889 and handle more seek cases correctly.
31891 2006-02-28 10:39:19 +0000 Wim Taymans <wim.taymans@gmail.com>
31893 gst/ffmpegcolorspace/gstffmpegcolorspace.c: Don't ignore return code from ffmpeg convert function.
31894 Original commit message from CVS:
31895 * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
31896 (gst_ffmpegcsp_transform):
31897 Don't ignore return code from ffmpeg convert function.
31898 * gst/ffmpegcolorspace/imgconvert.c: (img_convert):
31899 Split out some long statements to ease debugging.
31901 2006-02-27 12:08:22 +0000 Jan Schmidt <thaytan@mad.scientist.com>
31903 ext/libvisual/visual.c: Don't use gst_pad_use_fixed_caps, because it prevents downstream from being able to renegotia...
31904 Original commit message from CVS:
31905 * ext/libvisual/visual.c: (gst_visual_init),
31906 (gst_vis_src_negotiate), (get_buffer), (plugin_init):
31907 Don't use gst_pad_use_fixed_caps, because it prevents downstream from
31908 being able to renegotiate the size. Instead, use the negotiation
31909 algorithm from the goom plugin to pick an initial output caps.
31910 Also, allow theoretical libvisual plugins that might support non-GL
31911 output even if they also do GL.
31913 2006-02-26 21:05:46 +0000 Julien Moutte <julien@moutte.net>
31915 ext/libvisual/visual.c: Load only non GL plugins. Fix some memleaks and possible negotiation issues.
31916 Original commit message from CVS:
31917 2006-02-26 Julien MOUTTE <julien@moutte.net>
31918 * ext/libvisual/visual.c: (gst_visual_init),
31919 (gst_visual_src_setcaps), (get_buffer), (gst_visual_chain),
31920 (plugin_init): Load only non GL plugins. Fix some memleaks and
31921 possible negotiation issues.
31923 2006-02-24 23:19:44 +0000 Julien Moutte <julien@moutte.net>
31925 gst-libs/gst/tag/tag.h: Adding Annodex tags here.
31926 Original commit message from CVS:
31927 2006-02-25 Julien MOUTTE <julien@moutte.net>
31928 * gst-libs/gst/tag/tag.h: Adding Annodex tags here.
31930 2006-02-24 18:55:27 +0000 Michael Smith <msmith@xiph.org>
31932 gst/typefind/gsttypefindfunctions.c: Fix CMML type find function to not require a specific minor version of the CMML ...
31933 Original commit message from CVS:
31934 * gst/typefind/gsttypefindfunctions.c: (mpeg4_video_type_find),
31935 (cmml_type_find), (plugin_init):
31936 Fix CMML type find function to not require a specific minor version
31937 of the CMML header.
31938 Add an MPEG4 video elementary stream typefind function.
31940 2006-02-24 17:31:53 +0000 Michael Smith <msmith@xiph.org>
31942 ext/ogg/gstoggdemux.c: Annodex support in ogg demuxer. Doesn't do very much without the other annodex patches (to come).
31943 Original commit message from CVS:
31944 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_parse_skeleton_fishead),
31945 (gst_ogg_pad_parse_skeleton_fisbone), (gst_ogg_pad_query_convert),
31946 (gst_ogg_demux_chain_peer), (gst_ogg_pad_submit_packet),
31947 (gst_ogg_demux_perform_seek), (gst_ogg_demux_read_chain),
31948 (gst_ogg_demux_read_end_chain), (gst_ogg_demux_collect_chain_info),
31949 (gst_ogg_demux_change_state), (gst_annodex_granule_to_time):
31950 Annodex support in ogg demuxer. Doesn't do very much without the
31951 other annodex patches (to come).
31953 2006-02-24 16:21:34 +0000 Tim-Philipp Müller <tim@centricular.net>
31955 gst-libs/gst/riff/riff-media.c:
31956 Original commit message from CVS:
31957 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
31958 Pick up palette for MS video v1 (#327028, patch by:
31959 Fabrizio Gennari <fabrizio dot get at tiscali dot it>)
31961 2006-02-24 13:54:04 +0000 Tim-Philipp Müller <tim@centricular.net>
31963 gst/ffmpegcolorspace/gstffmpegcolorspace.c: The 'palette_data' field from incoming RGB caps shouldn't be proxied on o...
31964 Original commit message from CVS:
31965 * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
31966 (gst_ffmpegcsp_caps_remove_format_info),
31967 (gst_ffmpegcsp_get_unit_size):
31968 The 'palette_data' field from incoming RGB caps shouldn't be
31969 proxied on outgoing YUV caps; also, restrict unit size
31970 adjustment in case of paletted data only to the unit that
31971 actually has a palette. Fixes #330711.
31973 2006-02-24 12:18:14 +0000 Tim-Philipp Müller <tim@centricular.net>
31975 gst/ffmpegcolorspace/gstffmpegcolorspace.c: Plug some memory leaks.
31976 Original commit message from CVS:
31977 * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
31978 (gst_ffmpegcsp_transform_caps), (gst_ffmpegcsp_set_caps),
31979 (gst_ffmpegcsp_finalize), (gst_ffmpegcsp_class_init),
31980 (gst_ffmpegcsp_get_unit_size):
31981 Plug some memory leaks.
31983 2006-02-24 10:18:52 +0000 Tim-Philipp Müller <tim@centricular.net>
31985 sys/: Add some _CFLAGS and _LIBS that seem to be missing and/or required for Cygwin (see #317048).
31986 Original commit message from CVS:
31987 * sys/ximage/Makefile.am:
31988 * sys/xvimage/Makefile.am:
31989 Add some _CFLAGS and _LIBS that seem to be missing
31990 and/or required for Cygwin (see #317048).
31992 2006-02-24 00:07:18 +0000 Tim-Philipp Müller <tim@centricular.net>
31995 ChangeLog surgery: use UTF-8 encoding in stead of ISO-8859-15
31996 Original commit message from CVS:
31997 ChangeLog surgery: use UTF-8 encoding in stead of ISO-8859-15
31999 2006-02-22 18:46:46 +0000 Wim Taymans <wim.taymans@gmail.com>
32001 ext/alsa/gstalsasrc.c: Fix description as pointed out by caugier.
32002 Original commit message from CVS:
32003 * ext/alsa/gstalsasrc.c:
32004 Fix description as pointed out by caugier.
32006 2006-02-22 10:29:22 +0000 Edward Hervey <bilboed@bilboed.com>
32008 gst/typefind/gsttypefindfunctions.c: Better 3gp typefinding.
32009 Original commit message from CVS:
32010 Reviewed by : Edward Hervey <edward@fluendo.com>
32011 * gst/typefind/gsttypefindfunctions.c: (q3gp_type_find),
32013 Better 3gp typefinding.
32015 2006-02-21 12:16:16 +0000 Tim-Philipp Müller <tim@centricular.net>
32017 ext/gnomevfs/gstgnomevfssrc.c: Don't send EOS event here, the base class will send one for us.
32018 Original commit message from CVS:
32019 * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_create):
32020 Don't send EOS event here, the base class will send one for us.
32021 * gst/playback/gstplaybasebin.c: (prepare_output):
32022 Subpictures without video stream aren't allowed either.
32023 * gst/subparse/gstsubparse.c: (gst_subparse_type_find):
32024 Fix debug statement copy'n'paste-o.
32026 2006-02-21 12:05:18 +0000 Tim-Philipp Müller <tim@centricular.net>
32028 ext/alsa/gstalsamixer.c: Fix issues with mixer keeping state when muting/unmuting and when changing the volume whilst...
32029 Original commit message from CVS:
32030 * ext/alsa/gstalsamixer.c: (gst_alsa_mixer_get_volume):
32031 Fix issues with mixer keeping state when muting/unmuting
32032 and when changing the volume whilst muted (see #331763
32035 2006-02-20 18:27:06 +0000 Tim-Philipp Müller <tim@centricular.net>
32037 gst/subparse/gstsubparse.c: Set right caps given that we send escaped text. Also, honour <i></i>, <b></b> and <u></u>...
32038 Original commit message from CVS:
32039 * gst/subparse/gstsubparse.c: (subrip_unescape_formatting),
32040 (parse_subrip), (gst_sub_parse_format_autodetect):
32041 Set right caps given that we send escaped text. Also,
32042 honour <i></i>, <b></b> and <u></u> markers that can be found
32043 in .srt files (fixes #310202).
32045 2006-02-20 16:21:14 +0000 Tim-Philipp Müller <tim@centricular.net>
32047 gst-libs/gst/audio/mixerutils.c: Make order in which elements are tried more determinable.
32048 Original commit message from CVS:
32049 * gst-libs/gst/audio/mixerutils.c:
32050 (element_factory_rank_compare_func):
32051 Make order in which elements are tried more determinable.
32053 2006-02-20 15:57:51 +0000 Julien Moutte <julien@moutte.net>
32055 gst/playback/gstdecodebin.c: Make decodebin reusable by fixing remove_element_chain first and then introduce a cleane...
32056 Original commit message from CVS:
32057 * gst/playback/gstdecodebin.c: (get_our_ghost_pad),
32058 (remove_element_chain), (cleanup_decodebin),
32059 (gst_decode_bin_change_state): Make decodebin reusable by
32060 fixing remove_element_chain first and then introduce a
32061 cleaner in state change to ->NULL. (Closes #331678)
32062 ------------------------------------------------------
32064 2006-02-19 14:32:35 +0000 Wim Taymans <wim.taymans@gmail.com>
32066 ext/gnomevfs/gstgnomevfssink.c: use 0666 mask when creating files so umask gets applied correctly. Fixes #331295.
32067 Original commit message from CVS:
32068 * ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_open_file):
32069 use 0666 mask when creating files so umask gets applied
32070 correctly. Fixes #331295.
32072 2006-02-19 14:16:16 +0000 Tim-Philipp Müller <tim@centricular.net>
32074 gst/subparse/: Add very basic parser for SSA subtitle streams (as often found in matroska files).
32075 Original commit message from CVS:
32076 * gst/subparse/Makefile.am:
32077 * gst/subparse/gstssaparse.c: (gst_ssa_parse_base_init),
32078 (gst_ssa_parse_dispose), (gst_ssa_parse_init),
32079 (gst_ssa_parse_class_init), (gst_ssa_parse_src_event),
32080 (gst_ssa_parse_sink_event), (gst_ssa_parse_setcaps),
32081 (gst_ssa_parse_remove_override_codes), (gst_ssa_parse_parse_line),
32082 (gst_ssa_parse_chain), (gst_ssa_parse_change_state):
32083 * gst/subparse/gstssaparse.h:
32084 * gst/subparse/gstsubparse.c: (plugin_init):
32085 Add very basic parser for SSA subtitle streams (as often
32086 found in matroska files).
32088 2006-02-19 14:09:40 +0000 Tim-Philipp Müller <tim@centricular.net>
32090 gst/playback/gstdecodebin.c: That should be text/x-pango-markup, not text/x-pango-layout.
32091 Original commit message from CVS:
32092 * gst/playback/gstdecodebin.c: (mimetype_is_raw):
32093 That should be text/x-pango-markup, not text/x-pango-layout.
32095 2006-02-19 12:41:03 +0000 Julien Moutte <julien@moutte.net>
32097 ext/pango/gsttextoverlay.c: Polishing.
32098 Original commit message from CVS:
32099 2006-02-19 Julien MOUTTE <julien@moutte.net>
32100 * ext/pango/gsttextoverlay.c: (gst_text_overlay_finalize):
32103 2006-02-19 12:05:23 +0000 Julien Moutte <julien@moutte.net>
32105 ext/pango/gsttextoverlay.c: Fix state change deadlock.
32106 Original commit message from CVS:
32107 2006-02-19 Julien MOUTTE <julien@moutte.net>
32108 * ext/pango/gsttextoverlay.c: (gst_text_overlay_base_init),
32109 (gst_text_overlay_finalize), (gst_text_overlay_init),
32110 (gst_text_overlay_setcaps), (gst_text_overlay_src_event),
32111 (gst_text_overlay_render_text),
32112 (gst_text_overlay_text_pad_link),
32113 (gst_text_overlay_text_event), (gst_text_overlay_video_event),
32114 (gst_text_overlay_pop_text), (gst_text_overlay_text_chain),
32115 (gst_text_overlay_video_chain), (gst_text_overlay_change_state):
32116 Fix state change deadlock.
32118 2006-02-19 11:56:28 +0000 Julien Moutte <julien@moutte.net>
32120 ext/pango/gsttextoverlay.*: Fix seeking both for muxed formats and subtitles files.
32121 Original commit message from CVS:
32122 2006-02-19 Julien MOUTTE <julien@moutte.net>
32123 * ext/pango/gsttextoverlay.c: (gst_text_overlay_base_init),
32124 (gst_text_overlay_finalize), (gst_text_overlay_init),
32125 (gst_text_overlay_setcaps), (gst_text_overlay_src_event),
32126 (gst_text_overlay_render_text),
32127 (gst_text_overlay_text_pad_link),
32128 (gst_text_overlay_text_event), (gst_text_overlay_video_event),
32129 (gst_text_overlay_pop_text), (gst_text_overlay_text_chain),
32130 (gst_text_overlay_video_chain), (gst_text_overlay_change_state):
32131 * ext/pango/gsttextoverlay.h: Fix seeking both for muxed formats
32132 and subtitles files.
32134 2006-02-19 00:40:38 +0000 Julien Moutte <julien@moutte.net>
32136 gst/playback/gstdecodebin.c: pango layout should be considered as row.
32137 Original commit message from CVS:
32138 2006-02-19 Julien MOUTTE <julien@moutte.net>
32139 * gst/playback/gstdecodebin.c: (mimetype_is_raw): pango layout
32140 should be considered as row.
32142 2006-02-19 00:25:16 +0000 Julien Moutte <julien@moutte.net>
32144 gst/playback/gststreaminfo.*: Introduce language informations.
32145 Original commit message from CVS:
32146 2006-02-19 Julien MOUTTE <julien@moutte.net>
32147 * gst/playback/gststreaminfo.c: (gst_stream_type_get_type),
32149 * gst/playback/gststreaminfo.h: Introduce language informations.
32151 2006-02-18 22:41:31 +0000 Jan Schmidt <thaytan@mad.scientist.com>
32153 sys/: Set shared memory segments to be deleted as soon as we have attached, that way they get cleaned up automaticall...
32154 Original commit message from CVS:
32155 * sys/ximage/ximagesink.c: (gst_ximagesink_check_xshm_calls),
32156 (gst_ximagesink_ximage_new), (gst_ximagesink_ximage_destroy):
32157 * sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_destroy),
32158 (gst_xvimagesink_check_xshm_calls), (gst_xvimagesink_xvimage_new):
32159 Set shared memory segments to be deleted as soon as we have attached,
32160 that way they get cleaned up automatically if we crash.
32162 2006-02-18 19:53:48 +0000 Julien Moutte <julien@moutte.net>
32164 ext/pango/: Those functions are called with lock held.
32165 Original commit message from CVS:
32166 2006-02-18 Julien MOUTTE <julien@moutte.net>
32167 * ext/pango/gstclockoverlay.c: (gst_clock_overlay_get_text):
32168 * ext/pango/gsttimeoverlay.c: (gst_time_overlay_get_text): Those
32169 functions are called with lock held.
32171 2006-02-18 19:51:47 +0000 Julien Moutte <julien@moutte.net>
32175 Original commit message from CVS:
32178 2006-02-18 19:10:35 +0000 Julien Moutte <julien@moutte.net>
32180 ext/pango/gsttextoverlay.c: Refactoring of textoverlay without collectpads. This now supports sparse subtitles coming...
32181 Original commit message from CVS:
32182 2006-02-18 Julien MOUTTE <julien@moutte.net>
32183 * ext/pango/gsttextoverlay.c: (gst_text_overlay_base_init),
32184 (gst_text_overlay_finalize), (gst_text_overlay_init),
32185 (gst_text_overlay_setcaps), (gst_text_overlay_src_event),
32186 (gst_text_overlay_render_text),
32187 (gst_text_overlay_text_pad_link),
32188 (gst_text_overlay_text_pad_unlink),
32189 (gst_text_overlay_text_event),
32190 (gst_text_overlay_video_event), (gst_text_overlay_pop_text),
32191 (gst_text_overlay_text_chain), (gst_text_overlay_video_chain),
32192 (gst_text_overlay_change_state): Refactoring of textoverlay
32193 without collectpads. This now supports sparse subtitles coming
32194 from a demuxer instead of a sub file. Seeking is still broken
32195 though. Need to discuss with wtay some more on how to handle
32197 * ext/pango/gsttextoverlay.h:
32198 * gst/playback/gstplaybin.c: (setup_sinks): Support linking with
32199 subtitles coming from the demuxer.
32201 2006-02-17 19:31:12 +0000 Wim Taymans <wim.taymans@gmail.com>
32203 ext/vorbis/vorbisenc.c: Use some more scaling functions.
32204 Original commit message from CVS:
32205 * ext/vorbis/vorbisenc.c: (gst_vorbisenc_convert_src),
32206 (gst_vorbisenc_convert_sink):
32207 Use some more scaling functions.
32209 2006-02-17 16:12:11 +0000 Tim-Philipp Müller <tim@centricular.net>
32211 ext/cdparanoia/gstcdparanoiasrc.*: Add back 'transport-error' and 'uncorrected-error' signals and make them actually ...
32212 Original commit message from CVS:
32213 * ext/cdparanoia/gstcdparanoiasrc.c:
32214 (gst_cd_paranoia_src_class_init), (gst_cd_paranoia_dummy_callback),
32215 (gst_cd_paranoia_paranoia_callback),
32216 (gst_cd_paranoia_src_signal_is_being_watched),
32217 (gst_cd_paranoia_src_read_sector):
32218 * ext/cdparanoia/gstcdparanoiasrc.h:
32219 Add back 'transport-error' and 'uncorrected-error' signals and
32220 make them actually be fired when bad stuff happens (#319340).
32222 2006-02-17 14:07:01 +0000 Wim Taymans <wim.taymans@gmail.com>
32224 gst-libs/gst/audio/gstringbuffer.c: Small cleanups.
32225 Original commit message from CVS:
32226 * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_get_type),
32227 (gst_ring_buffer_open_device), (gst_ring_buffer_close_device),
32228 (gst_ring_buffer_device_is_open), (gst_ring_buffer_acquire),
32229 (gst_ring_buffer_release), (gst_ring_buffer_set_flushing),
32230 (gst_ring_buffer_start), (gst_ring_buffer_pause_unlocked),
32231 (gst_ring_buffer_pause), (gst_ring_buffer_stop),
32232 (gst_ring_buffer_delay), (gst_ring_buffer_samples_done),
32233 (gst_ring_buffer_set_sample), (gst_ring_buffer_clear_all),
32234 (gst_ring_buffer_commit), (gst_ring_buffer_prepare_read),
32235 (gst_ring_buffer_clear):
32237 Added some G_LIKELY.
32239 2006-02-17 10:15:52 +0000 Wim Taymans <wim.taymans@gmail.com>
32241 gst-libs/gst/audio/TODO: Update TODO
32242 Original commit message from CVS:
32243 * gst-libs/gst/audio/TODO:
32245 * gst-libs/gst/audio/gstbaseaudiosink.c:
32246 (gst_base_audio_sink_get_offset):
32247 When trying to play samples ASAP and we don't have a
32248 previous sample, try to play at position 0 instead of
32249 an invalid position.
32251 2006-02-17 09:24:56 +0000 Wim Taymans <wim.taymans@gmail.com>
32253 ext/alsa/gstalsasink.c: Also release lock when we get an error in _reset(); fix an error message.
32254 Original commit message from CVS:
32255 * ext/alsa/gstalsasink.c: (gst_alsasink_open),
32256 (gst_alsasink_reset):
32257 Also release lock when we get an error in _reset();
32258 fix an error message.
32260 2006-02-16 21:01:23 +0000 Tim-Philipp Müller <tim@centricular.net>
32262 ext/alsa/gstalsasink.*: Add support for more than 2 channels (#326720).
32263 Original commit message from CVS:
32264 * ext/alsa/gstalsasink.c: (gst_alsasink_class_init),
32265 (gst_alsasink_init), (get_channel_free_structure),
32266 (caps_add_channel_configuration), (gst_alsasink_getcaps),
32267 (gst_alsasink_close):
32268 * ext/alsa/gstalsasink.h:
32269 Add support for more than 2 channels (#326720).
32271 2006-02-16 20:19:51 +0000 Tim-Philipp Müller <tim@centricular.net>
32273 gst-libs/gst/riff/riff-media.c: Set codec_name for WAVEFORMATEX as well. When we have 'normal PCM' with 4 or 6 channe...
32274 Original commit message from CVS:
32275 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
32276 Set codec_name for WAVEFORMATEX as well. When we have 'normal PCM'
32277 with 4 or 6 channels, assume a default channel layout to make things
32278 work (not sure there's anything else we can do in those cases).
32280 2006-02-16 19:18:46 +0000 Tim-Philipp Müller <tim@centricular.net>
32282 gst-libs/gst/audio/multichannel.c: Minor docs fix.
32283 Original commit message from CVS:
32284 * gst-libs/gst/audio/multichannel.c:
32286 * gst-libs/gst/riff/Makefile.am:
32287 * gst-libs/gst/riff/riff-ids.h:
32288 * gst-libs/gst/riff/riff-media.c:
32289 (gst_riff_wavext_add_channel_layout), (gst_riff_create_audio_caps):
32290 Add support for WAVEFORMATEX, eg. PCM audio with more than two
32291 channels and a channel layout map.
32293 2006-02-16 17:06:46 +0000 Edward Hervey <bilboed@bilboed.com>
32295 gst/videoscale/vs_scanline.c: C-level optimization of the RGBA nearest neighbour function.
32296 Original commit message from CVS:
32297 Reviewed by Edward Hervey <edward@fluendo.com>
32298 * gst/videoscale/vs_scanline.c: (vs_scanline_resample_nearest_RGBA):
32299 C-level optimization of the RGBA nearest neighbour function.
32300 Eventually this might end up in liboil with vectorized versions.
32302 2006-02-16 11:44:43 +0000 Tim-Philipp Müller <tim@centricular.net>
32304 gst-libs/gst/audio/multichannel.c: When we have more than 2 channels, but no channel layout is specified in the caps,...
32305 Original commit message from CVS:
32306 * gst-libs/gst/audio/multichannel.c:
32307 (gst_audio_get_channel_positions):
32308 When we have more than 2 channels, but no channel layout is
32309 specified in the caps, return some default channel layout
32310 to the caller and warn about about a possibly buggy element
32311 (could be buggy filtercaps as well of course) (#317038).
32313 2006-02-16 09:29:38 +0000 Tim-Philipp Müller <tim@centricular.net>
32315 pkgconfig/gstreamer-plugins-base-uninstalled.pc.in: Add gst-libs/gst/cdda to list of lib search paths.
32316 Original commit message from CVS:
32317 * pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
32318 Add gst-libs/gst/cdda to list of lib search paths.
32320 2006-02-15 12:20:47 +0000 Andy Wingo <wingo@pobox.com>
32322 ext/ogg/gstoggmux.c (gst_ogg_mux_collected): When updating timestamp, update timestamp_end as well. Fixes a bugaboo. ...
32323 Original commit message from CVS:
32324 2006-02-15 Andy Wingo <wingo@pobox.com>
32325 * ext/ogg/gstoggmux.c (gst_ogg_mux_collected): When updating
32326 timestamp, update timestamp_end as well. Fixes a bugaboo. I hope
32327 to the Lord Jesus that I do not have to touch the ogg muxer ever
32330 2006-02-15 12:07:57 +0000 Edward Hervey <bilboed@bilboed.com>
32332 gst/typefind/gsttypefindfunctions.c: quicktime movie files can also contain 'uuid' atoms.
32333 Original commit message from CVS:
32334 * gst/typefind/gsttypefindfunctions.c: (qt_type_find):
32335 quicktime movie files can also contain 'uuid' atoms.
32337 2006-02-14 18:52:52 +0000 Tim-Philipp Müller <tim@centricular.net>
32339 gst/audioconvert/plugin.c: Register the GstAudioChannelPosition enum type with the type system in the plugin_init fun...
32340 Original commit message from CVS:
32341 * gst/audioconvert/plugin.c: (plugin_init):
32342 Register the GstAudioChannelPosition enum type with the type
32343 system in the plugin_init function, so that it is known before
32344 any element actually makes use of multi-channel stuff. This is
32345 required for example if one wants to be able to deserialise/use
32346 a caps string with channel positions before any pipeline has
32347 been setup and started, like with gst-launch.
32349 2006-02-14 13:45:35 +0000 Wim Taymans <wim.taymans@gmail.com>
32351 gst-libs/gst/audio/gstringbuffer.c: Add some compiler G_(UN_)LIKELY help.
32352 Original commit message from CVS:
32353 * gst-libs/gst/audio/gstringbuffer.c:
32354 (gst_ring_buffer_pause_unlocked), (gst_ring_buffer_delay),
32355 (gst_ring_buffer_samples_done), (wait_segment),
32356 (gst_ring_buffer_commit), (gst_ring_buffer_clear):
32357 Add some compiler G_(UN_)LIKELY help.
32358 SIGNAL the ringbuffer waiters when going to PAUSED as well to
32359 make sure they can exit their functions. Should fix #330748
32361 2006-02-13 20:49:07 +0000 Thomas Vander Stichele <thomas@apestaart.org>
32363 Windows does not have long long; copy the generated _stdint.h
32364 Original commit message from CVS:
32368 * win32/common/_stdint.h:
32369 Windows does not have long long; copy the generated _stdint.h
32370 * win32/common/interfaces-enumtypes.c:
32371 (gst_color_balance_type_get_type), (gst_mixer_type_get_type),
32372 (gst_mixer_track_flags_get_type),
32373 (gst_tuner_channel_flags_get_type):
32374 * win32/common/multichannel-enumtypes.c:
32375 (gst_audio_channel_position_get_type):
32378 2006-02-13 18:49:02 +0000 Wim Taymans <wim.taymans@gmail.com>
32380 gst-libs/gst/audio/gstbaseaudiosink.c: Always sync on first sample we receive when starting.
32381 Original commit message from CVS:
32382 * gst-libs/gst/audio/gstbaseaudiosink.c:
32383 (gst_base_audio_sink_get_time), (gst_base_audio_sink_preroll),
32384 (gst_base_audio_sink_render), (gst_base_audio_sink_change_state):
32385 Always sync on first sample we receive when starting.
32387 2006-02-13 15:59:48 +0000 Wim Taymans <wim.taymans@gmail.com>
32389 gst/playback/gstplaybin.c: Update vis bin docs.
32390 Original commit message from CVS:
32391 * gst/playback/gstplaybin.c: (gen_vis_element):
32392 Update vis bin docs.
32393 Move queue after tee so we don't queue video buffers but
32394 audio samples instead. Fixes problems where the video queue
32395 is filled and the audio queue empty.
32397 2006-02-13 15:17:34 +0000 Tim-Philipp Müller <tim@centricular.net>
32399 gst-libs/gst/cdda/gstcddabasesrc.c: No need to push an EOS event here, GstBaseSrc will do that for us when we return ...
32400 Original commit message from CVS:
32401 * gst-libs/gst/cdda/gstcddabasesrc.c: (gst_cdda_base_src_create):
32402 No need to push an EOS event here, GstBaseSrc will do that for us
32403 when we return FLOW_UNEXPECTED.
32405 2006-02-12 14:54:55 +0000 Wim Taymans <wim.taymans@gmail.com>
32407 gst-libs/gst/audio/gstbaseaudiosink.c: Use scale functions when possible.
32408 Original commit message from CVS:
32409 * gst-libs/gst/audio/gstbaseaudiosink.c:
32410 (gst_base_audio_sink_get_time), (gst_base_audio_sink_setcaps),
32411 (gst_base_audio_sink_drain), (gst_base_audio_sink_preroll),
32412 (gst_base_audio_sink_render), (gst_base_audio_sink_change_state):
32413 Use scale functions when possible.
32414 Fix error messages.
32415 Free clockid when after waiting for EOS.
32416 Use G_(UN_)LIKLY when it makes sense.
32417 Fix sample clipping bug found by Arwed v. Merkatz fixes #330789.
32419 2006-02-12 14:26:55 +0000 Edward Hervey <bilboed@bilboed.com>
32421 gst/playback/gstplaybasebin.c: Remove stray semi-colon (fixes #330888).
32422 Original commit message from CVS:
32423 * gst/playback/gstplaybasebin.c: (prepare_output):
32424 Remove stray semi-colon (fixes #330888).
32426 2006-02-11 23:35:55 +0000 Jan Schmidt <thaytan@mad.scientist.com>
32428 sys/: Fix up the XShm call testing so that we catch errors, and don't cause new ones by attempting to detach from a s...
32429 Original commit message from CVS:
32430 * sys/ximage/ximagesink.c: (gst_ximagesink_check_xshm_calls):
32431 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_check_xshm_calls):
32432 Fix up the XShm call testing so that we catch errors, and don't
32433 cause new ones by attempting to detach from a segment we failed
32434 to attach to. Fixes #312439.
32436 2006-02-10 11:29:55 +0000 Edward Hervey <bilboed@bilboed.com>
32438 gst/typefind/gsttypefindfunctions.c: Added flv file typefind (video/x-flv).
32439 Original commit message from CVS:
32440 * gst/typefind/gsttypefindfunctions.c: (plugin_init):
32441 Added flv file typefind (video/x-flv).
32443 2006-02-10 10:53:33 +0000 Edward Hervey <bilboed@bilboed.com>
32445 gst-libs/gst/riff/riff-media.c: Added FLV1 <==> 'video/x-flash-video,flvversion=1' conversion.
32446 Original commit message from CVS:
32447 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
32448 (gst_riff_create_video_template_caps):
32449 Added FLV1 <==> 'video/x-flash-video,flvversion=1' conversion.
32450 Also added the caps to the default set of riff video caps.
32452 2006-02-09 19:05:23 +0000 Andy Wingo <wingo@pobox.com>
32454 ext/ogg/gstoggmux.c (GstOggPad): Keep track of both the start time and the end time of the last packet in the page.
32455 Original commit message from CVS:
32456 2006-02-09 Andy Wingo <wingo@pobox.com>
32457 * ext/ogg/gstoggmux.c (GstOggPad): Keep track of both the start
32458 time and the end time of the last packet in the page.
32459 (gst_ogg_mux_pad_queue_page): In addition to setting the timestamp
32460 on the pages in our queue, set the duration as well. Reflow a
32462 (gst_ogg_mux_collected): Keep track of GstOggPad->timestamp_end.
32463 Fixes bad muxing order.
32465 2006-02-09 17:04:18 +0000 Thomas Vander Stichele <thomas@apestaart.org>
32467 gst-libs/gst/rtp/gstbasertppayload.c: update seqnum before setting it on the packet; this makes sure that the timesta...
32468 Original commit message from CVS:
32469 * gst-libs/gst/rtp/gstbasertppayload.c:
32470 (gst_basertppayload_setcaps), (gst_basertppayload_push):
32471 update seqnum before setting it on the packet; this makes sure
32472 that the timestamp and seqnum properties match after pushing
32475 2006-02-09 12:16:35 +0000 Andy Wingo <wingo@pobox.com>
32479 Original commit message from CVS:
32482 2006-02-09 11:46:03 +0000 Andy Wingo <wingo@pobox.com>
32484 * gst-libs/gst/audio/gstringbuffer.c:
32485 * win32/common/config.h:
32487 Original commit message from CVS:
32490 2006-02-09 11:36:18 +0000 Andy Wingo <wingo@pobox.com>
32492 gst-libs/gst/audio/gstringbuffer.c
32493 Original commit message from CVS:
32494 2006-02-09 Andy Wingo <wingo@pobox.com>
32495 * gst-libs/gst/audio/gstringbuffer.c
32496 (gst_ring_buffer_samples_done): Cast to guint64, fixes an integer
32497 overflow after 13.5 hours of recording. Kapow!
32498 * ext/alsa/gstalsasrc.c (gst_alsasrc_delay): Clamp the delay to
32499 the buffer size -- we don't care about underrun/overrun reporting
32500 right now, just need to return a useful value.
32502 2006-02-09 11:21:33 +0000 Jan Schmidt <thaytan@mad.scientist.com>
32504 configure.ac: Back to CVS
32505 Original commit message from CVS:
32509 === release 0.10.3 ===
32511 2006-02-09 11:18:22 +0000 Jan Schmidt <thaytan@mad.scientist.com>
32517 * docs/plugins/inspect/plugin-adder.xml:
32518 * docs/plugins/inspect/plugin-alsa.xml:
32519 * docs/plugins/inspect/plugin-audioconvert.xml:
32520 * docs/plugins/inspect/plugin-audiorate.xml:
32521 * docs/plugins/inspect/plugin-audioresample.xml:
32522 * docs/plugins/inspect/plugin-audiotestsrc.xml:
32523 * docs/plugins/inspect/plugin-cdparanoia.xml:
32524 * docs/plugins/inspect/plugin-decodebin.xml:
32525 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
32526 * docs/plugins/inspect/plugin-gnomevfs.xml:
32527 * docs/plugins/inspect/plugin-libvisual.xml:
32528 * docs/plugins/inspect/plugin-ogg.xml:
32529 * docs/plugins/inspect/plugin-pango.xml:
32530 * docs/plugins/inspect/plugin-playbin.xml:
32531 * docs/plugins/inspect/plugin-subparse.xml:
32532 * docs/plugins/inspect/plugin-tcp.xml:
32533 * docs/plugins/inspect/plugin-theora.xml:
32534 * docs/plugins/inspect/plugin-typefindfunctions.xml:
32535 * docs/plugins/inspect/plugin-video4linux.xml:
32536 * docs/plugins/inspect/plugin-videorate.xml:
32537 * docs/plugins/inspect/plugin-videoscale.xml:
32538 * docs/plugins/inspect/plugin-videotestsrc.xml:
32539 * docs/plugins/inspect/plugin-volume.xml:
32540 * docs/plugins/inspect/plugin-vorbis.xml:
32541 * docs/plugins/inspect/plugin-ximagesink.xml:
32542 * docs/plugins/inspect/plugin-xvimagesink.xml:
32543 * win32/common/config.h:
32545 Original commit message from CVS:
32548 2006-02-08 18:37:38 +0000 Jan Schmidt <thaytan@mad.scientist.com>
32550 configure.ac: Drat. Bump libtool version number for new API.
32551 Original commit message from CVS:
32553 Drat. Bump libtool version number for new API.
32554 Prelease 0.10.2.3 (of 0.10.3)
32556 2006-02-08 15:57:53 +0000 Jan Schmidt <thaytan@mad.scientist.com>
32558 0.10.2.2 prerelease (of 0.10.3).
32559 Original commit message from CVS:
32561 * win32/common/config.h:
32562 0.10.2.2 prerelease (of 0.10.3).
32564 2006-02-08 15:50:08 +0000 Jan Schmidt <thaytan@mad.scientist.com>
32566 gst/tcp/gsttcpclientsrc.c: Revert Andy's newsegment change pending a more correct fix.
32567 Original commit message from CVS:
32568 * gst/tcp/gsttcpclientsrc.c: (gst_tcp_client_src_create):
32569 Revert Andy's newsegment change pending a more correct
32572 2006-02-08 12:46:14 +0000 Jan Schmidt <thaytan@mad.scientist.com>
32589 Original commit message from CVS:
32592 2006-02-08 11:04:09 +0000 Thomas Vander Stichele <thomas@apestaart.org>
32594 * gst/tcp/gstmultifdsink.c:
32596 Original commit message from CVS:
32599 2006-02-08 09:20:23 +0000 Thomas Vander Stichele <thomas@apestaart.org>
32601 gst/typefind/gsttypefindfunctions.c: detect more files as 3gp group and reorder the iso file formats
32602 Original commit message from CVS:
32604 * gst/typefind/gsttypefindfunctions.c: (q3gp_type_find),
32605 (qt_type_find), (plugin_init):
32606 detect more files as 3gp
32607 group and reorder the iso file formats
32609 2006-02-07 18:32:00 +0000 Tim-Philipp Müller <tim@centricular.net>
32611 ext/vorbis/vorbis.c: Register musicbrainz tags, so apps don't have to.
32612 Original commit message from CVS:
32613 * ext/vorbis/vorbis.c: (plugin_init):
32614 Register musicbrainz tags, so apps don't have to.
32616 2006-02-07 17:44:53 +0000 Tim-Philipp Müller <tim@centricular.net>
32618 gst-libs/gst/tag/gstvorbistag.c: Make sure we called gst_tag_register_musicbrainz_tags() before possibly mapping a vo...
32619 Original commit message from CVS:
32620 * gst-libs/gst/tag/gstvorbistag.c: (gst_tag_from_vorbis_tag),
32621 (gst_tag_to_vorbis_tag):
32622 Make sure we called gst_tag_register_musicbrainz_tags()
32623 before possibly mapping a vorbiscomment string from/to a
32626 2006-02-07 16:16:41 +0000 Tim-Philipp Müller <tim@centricular.net>
32628 gst/typefind/gsttypefindfunctions.c: In case we can't find the required number of consecutive mpeg audio frames to po...
32629 Original commit message from CVS:
32630 * gst/typefind/gsttypefindfunctions.c: (mp3_type_find):
32631 In case we can't find the required number of consecutive
32632 mpeg audio frames to positively identify an MPEG audio
32633 stream, check if there's at least a valid mpeg audio
32634 frame right at offset 0 and if so suggest mpeg/audio
32635 caps with a very low probability (#153004).
32637 2006-02-07 15:52:26 +0000 Andy Wingo <wingo@pobox.com>
32639 gst/tcp/gsttcpclientsrc.c (gst_tcp_client_src_create): Switch to a TIME segment if we get timestamped buffers. Requir...
32640 Original commit message from CVS:
32641 2006-02-07 Andy Wingo <wingo@pobox.com>
32642 * gst/tcp/gsttcpclientsrc.c (gst_tcp_client_src_create): Switch to
32643 a TIME segment if we get timestamped buffers. Requires recent
32644 fixes in core to work properly.
32646 2006-02-07 14:57:46 +0000 Tim-Philipp Müller <tim@centricular.net>
32648 gst/playback/gstplaybasebin.c: Don't print the URI as part of the error message, it makes error dialogs look rather u...
32649 Original commit message from CVS:
32650 * gst/playback/gstplaybasebin.c: (prepare_output):
32651 Don't print the URI as part of the error message, it
32652 makes error dialogs look rather ugly, especially if
32653 the URI is very long or has characters in it that
32656 2006-02-07 13:11:31 +0000 Tim-Philipp Müller <tim@centricular.net>
32658 gst/playback/gstplaybasebin.c: Error out if we have only text or subtitles, but nothing else. Also error out if we ha...
32659 Original commit message from CVS:
32660 * gst/playback/gstplaybasebin.c: (prepare_output):
32661 Error out if we have only text or subtitles, but nothing
32662 else. Also error out if we have subtitles but no video
32665 2006-02-07 11:44:39 +0000 Tim-Philipp Müller <tim@centricular.net>
32667 ext/gnomevfs/gstgnomevfssrc.c: Treat GNOME_VFS_RESULT_EOF as EOS, not as error (#329194).
32668 Original commit message from CVS:
32669 * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_create):
32670 Treat GNOME_VFS_RESULT_EOF as EOS, not as error (#329194).
32671 Post an error message on the bus when we encounter an
32672 error, which will hopefully be more meaningful than the
32673 'Internal Flow Error' message users get to see if we
32674 just return GST_FLOW_ERROR.
32676 2006-02-07 11:28:04 +0000 Andy Wingo <wingo@pobox.com>
32678 configure.ac (GST_MAJORMINOR): Update core version req to 0.10.2.2, for the collectpads API addition (#330244).
32679 Original commit message from CVS:
32680 2006-02-07 Andy Wingo <wingo@pobox.com>
32681 * configure.ac (GST_MAJORMINOR): Update core version req to
32682 0.10.2.2, for the collectpads API addition (#330244).
32684 2006-02-06 19:09:26 +0000 Tim-Philipp Müller <tim@centricular.net>
32686 ext/gnomevfs/gstgnomevfs.c: Return FALSE from plugin_init() when GnomeVFS can't be initialised for some reason (#3284...
32687 Original commit message from CVS:
32688 * ext/gnomevfs/gstgnomevfs.c: (plugin_init):
32689 Return FALSE from plugin_init() when GnomeVFS can't
32690 be initialised for some reason (#328423).
32692 2006-02-06 13:26:54 +0000 Julien Moutte <julien@moutte.net>
32694 ext/pango/gsttextoverlay.c: Stick to seeking theory until i find the bug.
32695 Original commit message from CVS:
32696 2006-02-06 Julien MOUTTE <julien@moutte.net>
32697 * ext/pango/gsttextoverlay.c: (gst_text_overlay_src_event):
32698 Stick to seeking theory until i find the bug.
32699 * gst/subparse/gstsubparse.c: (parse_subrip): Fix debug.
32701 2006-02-06 12:38:48 +0000 Jan Schmidt <thaytan@mad.scientist.com>
32703 Make theoraenc and the tests leak free. Like, really.
32704 Original commit message from CVS:
32705 * ext/theora/theoraenc.c: (gst_theora_enc_class_init),
32706 (theora_enc_finalize), (theora_enc_sink_setcaps),
32707 (theora_set_header_on_caps), (theora_enc_chain),
32708 (theora_enc_change_state):
32709 * tests/check/pipelines/theoraenc.c: (GST_START_TEST):
32710 Make theoraenc and the tests leak free. Like, really.
32712 2006-02-05 23:31:05 +0000 Jan Schmidt <thaytan@mad.scientist.com>
32714 Add a finalize method to ensure we clean up state even if someone omitted the state change back to NULL.
32715 Original commit message from CVS:
32716 (theora_enc_finalize), (theora_enc_sink_setcaps):
32717 Add a finalize method to ensure we clean up state even if
32718 someone omitted the state change back to NULL.
32719 * ext/vorbis/vorbisenc.c: (gst_vorbisenc_metadata_set1),
32720 (gst_vorbisenc_chain):
32721 Free some more leaked bits.
32722 * tests/check/pipelines/theoraenc.c: (start_pipeline),
32724 Wait for state changes to happen if they're ASYNC.
32725 This ought to teach those fancy pants buildbots a lesson.
32727 2006-02-05 22:47:41 +0000 Jan Schmidt <thaytan@mad.scientist.com>
32729 gst-libs/gst/tag/gstid3tag.c: Add mapping for ID3 International Standard Recording Code tag "TSRC"
32730 Original commit message from CVS:
32731 * gst-libs/gst/tag/gstid3tag.c:
32732 Add mapping for ID3 International Standard Recording Code
32735 2006-02-05 22:44:55 +0000 Jan Schmidt <thaytan@mad.scientist.com>
32737 ext/vorbis/vorbisenc.c: Don't leak tag names.
32738 Original commit message from CVS:
32739 * ext/vorbis/vorbisenc.c: (gst_vorbisenc_metadata_set1):
32740 Don't leak tag names.
32742 2006-02-05 18:22:01 +0000 Tim-Philipp Müller <tim@centricular.net>
32744 Split libgsttag docs into multiple sections.
32745 Original commit message from CVS:
32746 * docs/libs/gst-plugins-base-libs-docs.sgml:
32747 * docs/libs/gst-plugins-base-libs-sections.txt:
32748 * gst-libs/gst/tag/gstid3tag.c:
32749 * gst-libs/gst/tag/gstvorbistag.c:
32750 * gst-libs/gst/tag/tags.c:
32751 Split libgsttag docs into multiple sections.
32753 2006-02-05 18:01:33 +0000 Tim-Philipp Müller <tim@centricular.net>
32755 Add libgsttag to the docs.
32756 Original commit message from CVS:
32757 * docs/libs/Makefile.am:
32758 * docs/libs/gst-plugins-base-libs-docs.sgml:
32759 * docs/libs/gst-plugins-base-libs-sections.txt:
32760 * gst-libs/gst/tag/gstid3tag.c: (gst_tag_from_id3_tag):
32761 * gst-libs/gst/tag/gstvorbistag.c:
32762 * gst-libs/gst/tag/tag.h:
32763 * gst-libs/gst/tag/tags.c:
32764 Add libgsttag to the docs.
32766 2006-02-05 17:21:23 +0000 Julien Moutte <julien@moutte.net>
32768 ext/pango/gsttextoverlay.c: Fix clockoverlay.
32769 Original commit message from CVS:
32770 2006-02-05 Julien MOUTTE <julien@moutte.net>
32771 * ext/pango/gsttextoverlay.c: (gst_text_overlay_finalize),
32772 (gst_text_overlay_init), (gst_text_overlay_src_event),
32773 (gst_text_overlay_collected): Fix clockoverlay.
32775 2006-02-05 17:15:17 +0000 Tim-Philipp Müller <tim@centricular.net>
32777 docs/libs/compiling.sgml: Fix typo: it's pkg-config, not pkg-gconfig
32778 Original commit message from CVS:
32779 * docs/libs/compiling.sgml:
32780 Fix typo: it's pkg-config, not pkg-gconfig
32781 * docs/libs/gst-plugins-base-libs-docs.sgml:
32782 * docs/libs/gst-plugins-base-libs-sections.txt:
32783 * docs/libs/tmpl/gstgconf.sgml:
32784 There is no libgstgconf in 0.10, remove it
32787 2006-02-05 16:03:48 +0000 Julien Moutte <julien@moutte.net>
32789 docs/libs/tmpl/gstcolorbalance.sgml: Updated.
32790 Original commit message from CVS:
32791 2006-02-05 Julien MOUTTE <julien@moutte.net>
32792 * docs/libs/tmpl/gstcolorbalance.sgml: Updated.
32793 * ext/pango/gsttextoverlay.c: (gst_text_overlay_init),
32794 (gst_text_overlay_src_event), (gst_text_overlay_collected):
32795 * gst/subparse/gstsubparse.c: (gst_sub_parse_dispose),
32796 (gst_sub_parse_class_init), (gst_sub_parse_init),
32797 (gst_sub_parse_src_event), (parse_mdvdsub), (parse_subrip),
32798 (parse_mpsub), (parser_state_init), (handle_buffer),
32799 (gst_sub_parse_chain), (gst_sub_parse_sink_event),
32801 * gst/subparse/gstsubparse.h: Introduce seeking code.
32803 2006-02-05 15:14:06 +0000 Tim-Philipp Müller <tim@centricular.net>
32805 gst-libs/gst/tag/gstvorbistag.c: Add comment about LANGUAGE tag inconsistency (we want
32806 Original commit message from CVS:
32807 * gst-libs/gst/tag/gstvorbistag.c: (gst_vorbis_tag_add):
32808 Add comment about LANGUAGE tag inconsistency (we want
32809 ISO-639-1, but extract three-letter identifiers?)
32811 Add two translatable files.
32813 2006-02-05 14:59:28 +0000 Tim-Philipp Müller <tim@centricular.net>
32815 gst-libs/gst/tag/: Forward-port some tags stuff from the 0.8 branch. This is mostly the addition of musicbrainz tags ...
32816 Original commit message from CVS:
32817 * gst-libs/gst/tag/Makefile.am:
32818 * gst-libs/gst/tag/gstvorbistag.c: (gst_vorbis_tag_add):
32819 * gst-libs/gst/tag/tag.h:
32820 * gst-libs/gst/tag/tags.c:
32821 (gst_tag_register_musicbrainz_tags_internal),
32822 (gst_tag_register_musicbrainz_tags):
32823 Forward-port some tags stuff from the 0.8 branch. This is
32824 mostly the addition of musicbrainz tags and their mapping
32825 to vorbistags, and a vorbistag mapping of the language tag.
32827 2006-02-05 12:06:25 +0000 Julien Moutte <julien@moutte.net>
32829 gst/playback/gstplaybin.c: Fix broken code refactoring.
32830 Original commit message from CVS:
32831 2006-02-05 Julien MOUTTE <julien@moutte.net>
32832 * gst/playback/gstplaybin.c: (gen_text_element): Fix broken code
32835 2006-02-05 03:05:41 +0000 David Schleef <ds@schleef.org>
32837 Add Dirac typefinding and add dirac format to oggmux.
32838 Original commit message from CVS:
32839 * ext/ogg/gstoggmux.c:
32840 * gst/typefind/gsttypefindfunctions.c:
32841 Add Dirac typefinding and add dirac format to oggmux.
32843 2006-02-04 07:49:03 +0000 Michael Smith <msmith@xiph.org>
32846 Improve error message for liboil missingness.
32847 Original commit message from CVS:
32848 Improve error message for liboil missingness.
32850 2006-02-03 19:23:41 +0000 Tim-Philipp Müller <tim@centricular.net>
32852 gst/playback/gstdecodebin.c: Don't put essential function call into g_return_*() macro, otherwise it'll all be replac...
32853 Original commit message from CVS:
32854 * gst/playback/gstdecodebin.c: (try_to_link_1):
32855 Don't put essential function call into
32856 g_return_*() macro, otherwise it'll all be
32857 replaced by NOOPs when compiling with
32858 G_DISABLE_CHECKS defined.
32860 2006-02-03 17:45:44 +0000 Edgard Lima <edgard.lima@indt.org.br>
32863 * ext/ogg/gstoggdemux.c:
32864 * ext/ogg/gstoggparse.c:
32865 * gst/tcp/gsttcpserversink.c:
32866 * sys/v4l/v4lsrc_calls.c:
32867 * sys/v4l/v4lsrc_calls.h:
32868 Just make it compile with --disable-gst-debug.
32869 Original commit message from CVS:
32870 Just make it compile with --disable-gst-debug.
32872 2006-02-03 12:51:47 +0000 Wim Taymans <wim.taymans@gmail.com>
32874 ext/alsa/gstalsasink.*: Add lock to protect alsa calls.
32875 Original commit message from CVS:
32876 * ext/alsa/gstalsasink.c: (gst_alsasink_finalise),
32877 (gst_alsasink_class_init), (gst_alsasink_init),
32878 (gst_alsasink_write), (gst_alsasink_reset):
32879 * ext/alsa/gstalsasink.h:
32880 Add lock to protect alsa calls.
32881 Implement reset to flush samples ASAP, does not work
32884 2006-02-02 18:18:31 +0000 Wim Taymans <wim.taymans@gmail.com>
32886 gst-libs/gst/audio/gstbaseaudiosink.c: Ugh.. getting late I guess...
32887 Original commit message from CVS:
32888 * gst-libs/gst/audio/gstbaseaudiosink.c:
32889 (gst_base_audio_sink_provide_clock):
32890 Ugh.. getting late I guess...
32892 2006-02-02 18:13:26 +0000 Wim Taymans <wim.taymans@gmail.com>
32894 gst-libs/gst/audio/gstbaseaudiosink.c: Don't try to provide a clock when we are not negotiated since we might not be ...
32895 Original commit message from CVS:
32896 * gst-libs/gst/audio/gstbaseaudiosink.c:
32897 (gst_base_audio_sink_provide_clock),
32898 (gst_base_audio_sink_set_property),
32899 (gst_base_audio_sink_get_property), (gst_base_audio_sink_render):
32900 Don't try to provide a clock when we are not negotiated since
32901 we might not be able to make it run.
32903 2006-02-02 17:51:48 +0000 Tim-Philipp Müller <tim@centricular.net>
32905 gst/playback/gstdecodebin.c: Unlinking two source pads is ... hard.
32906 Original commit message from CVS:
32907 * gst/playback/gstdecodebin.c: (try_to_link_1):
32908 Unlinking two source pads is ... hard.
32910 2006-02-02 12:14:35 +0000 Wim Taymans <wim.taymans@gmail.com>
32912 gst-libs/gst/audio/TODO: Updated.
32913 Original commit message from CVS:
32914 * gst-libs/gst/audio/TODO:
32916 * gst-libs/gst/audio/gstbaseaudiosink.c:
32917 (gst_base_audio_sink_drain), (gst_base_audio_sink_event):
32918 On EOS, wait till the last sample is played before posting EOS.
32920 2006-02-02 08:53:27 +0000 Thomas Vander Stichele <thomas@apestaart.org>
32922 * tests/check/pipelines/theoraenc.c:
32923 comment on my understanding
32924 Original commit message from CVS:
32925 comment on my understanding
32927 2006-02-02 08:47:42 +0000 Thomas Vander Stichele <thomas@apestaart.org>
32930 * tests/check/pipelines/theoraenc.c:
32931 reformat to fit 80 chars
32932 Original commit message from CVS:
32933 reformat to fit 80 chars
32935 2006-02-02 00:04:37 +0000 Kai Vehmanen <kv2004@eca.cx>
32937 gst-libs/gst/rtp/gstbasertpdepayload.c: setting queue_delay to zero. Also avoid thread being started if queue_delay i...
32938 Original commit message from CVS:
32939 2006-02-01 Philippe Kalaf <burger at speedy dot org>
32940 * gst-libs/gst/rtp/gstbasertpdepayload.c:
32941 Patch by Kai Vehmanen : Adds ability to enable newsegment bypass by
32942 setting queue_delay to zero. Also avoid thread being started if
32943 queue_delay is zero.
32945 2006-02-01 14:51:29 +0000 Tim-Philipp Müller <tim@centricular.net>
32947 gst/playback/test6.c: Make test work again by connecting fakesinks to each decoded pad, which makes the pipeline wait...
32948 Original commit message from CVS:
32949 * gst/playback/test6.c: (new_decoded_pad_cb), (show_error), (main):
32950 Make test work again by connecting fakesinks to each decoded pad,
32951 which makes the pipeline wait until each fakesink has a buffer
32952 queued before going to PAUSED state. At that point we know the
32953 decodebin pads are negotiated.
32955 2006-02-01 11:59:47 +0000 Tim-Philipp Müller <tim@centricular.net>
32957 gst/: Pass unhandled queries to the parent class's query function.
32958 Original commit message from CVS:
32959 * gst-libs/gst/cdda/gstcddabasesrc.c: (gst_cdda_base_src_query),
32960 (gst_cdda_base_src_handle_event):
32961 * gst/audiotestsrc/gstaudiotestsrc.c: (gst_audio_test_src_query):
32962 Pass unhandled queries to the parent class's query function.
32964 2006-02-01 11:56:11 +0000 Tim-Philipp Müller <tim@centricular.net>
32966 Pass unhandled queries upstream instead of just dropping them (#326447). Also, fix supported query types list for som...
32967 Original commit message from CVS:
32968 * ext/ogg/gstoggdemux.c: (gst_ogg_pad_query_types),
32969 (gst_ogg_pad_src_query):
32970 * ext/ogg/gstogmparse.c: (gst_ogm_parse_sink_query):
32971 * ext/theora/theoradec.c: (theora_dec_src_query),
32972 (theora_dec_sink_query):
32973 * ext/vorbis/vorbisdec.c: (vorbis_dec_src_query),
32974 (vorbis_dec_sink_query):
32975 * ext/vorbis/vorbisenc.c: (gst_vorbisenc_src_query),
32976 (gst_vorbisenc_sink_query):
32977 * gst/adder/gstadder.c: (gst_adder_query):
32978 Pass unhandled queries upstream instead of just
32979 dropping them (#326447). Also, fix supported
32980 query types list for some elements.
32982 2006-02-01 09:58:15 +0000 Tim-Philipp Müller <tim@centricular.net>
32984 gst/typefind/gsttypefindfunctions.c: Fix typefinding for audio/x-au, audio/x-paris and audio/iLBC-sh. We cannot use t...
32985 Original commit message from CVS:
32986 * gst/typefind/gsttypefindfunctions.c: (au_type_find),
32987 (paris_type_find), (ilbc_type_find), (plugin_init):
32988 Fix typefinding for audio/x-au, audio/x-paris and
32989 audio/iLBC-sh. We cannot use the START_WITH macros
32990 here, because there can only be one typefind factory
32991 with the same name (caps), so the second one would
32992 replace the first one and the first one would never
32993 be called when doing typefinding (see #161712).
32995 2006-01-31 19:25:10 +0000 Wim Taymans <wim.taymans@gmail.com>
32997 ext/vorbis/vorbisdec.c: Use scale_int when we can, add some more scaling.
32998 Original commit message from CVS:
32999 * ext/vorbis/vorbisdec.c: (vorbis_dec_convert),
33000 (vorbis_handle_header_packet), (vorbis_dec_push),
33001 (vorbis_handle_data_packet):
33002 Use scale_int when we can, add some more scaling.
33003 Check packettype before parsing it.
33005 2006-01-31 17:44:35 +0000 Wim Taymans <wim.taymans@gmail.com>
33007 ext/theora/theoradec.c: Call right _scale functions.
33008 Original commit message from CVS:
33009 * ext/theora/theoradec.c: (_theora_granule_time),
33010 (theora_dec_src_convert), (theora_dec_sink_convert):
33011 Call right _scale functions.
33012 Use parameter instead of some other random value.
33014 2006-01-31 17:27:00 +0000 Wim Taymans <wim.taymans@gmail.com>
33016 ext/theora/theoradec.c: Use higher precision timestamps calculation.
33017 Original commit message from CVS:
33018 * ext/theora/theoradec.c: (_theora_granule_frame),
33019 (_theora_granule_time), (_inc_granulepos),
33020 (theora_dec_src_convert), (theora_dec_sink_convert),
33021 (theora_handle_type_packet), (theora_handle_data_packet),
33022 (theora_dec_chain):
33023 Use higher precision timestamps calculation.
33024 Convert some other conversions to _scale.
33026 2006-01-31 17:19:09 +0000 Stefan Kost <ensonic@users.sourceforge.net>
33028 gst/: initialize gst_controller before using
33029 Original commit message from CVS:
33030 * gst/audiotestsrc/gstaudiotestsrc.c:
33031 (gst_audio_test_src_create_sine_table), (plugin_init):
33032 * gst/volume/gstvolume.c: (plugin_init):
33033 initialize gst_controller before using
33035 2006-01-31 16:26:57 +0000 Jan Schmidt <thaytan@mad.scientist.com>
33037 tests/check/pipelines/: Define constant using G_GINT64_CONSTANT to avoid errors when passing it around - otherwise it...
33038 Original commit message from CVS:
33039 * tests/check/pipelines/theoraenc.c:
33040 * tests/check/pipelines/vorbisenc.c:
33041 Define constant using G_GINT64_CONSTANT to avoid errors when
33042 passing it around - otherwise it gets truncated to 32 bits.
33043 Fixes failing tests.
33045 2006-01-31 15:36:13 +0000 Andy Wingo <wingo@pobox.com>
33047 sys/v4l/gstv4lsrc.c (gst_v4lsrc_set_caps): Don't segfault if the caps being set doesn't have a framerate value. Basic...
33048 Original commit message from CVS:
33049 2006-01-31 Andy Wingo <wingo@pobox.com>
33050 * sys/v4l/gstv4lsrc.c (gst_v4lsrc_set_caps): Don't segfault if the
33051 caps being set doesn't have a framerate value. Basically a stopgap
33053 * ext/ogg/gstoggmux.c (GST_BUFFER_END_TIME): New macro. Not
33054 technically correct enough to put into core though.
33055 (gst_ogg_mux_dequeue_page): Use END_TIME instead of TIMESTAMP +
33056 DURATION. Fixes theoraenc ! oggmux.
33057 * sys/v4l/gstv4lsrc.c (gst_v4lsrc_fixate): Fixate to the nearest
33058 fraction, not double.
33060 2006-01-31 12:23:35 +0000 Christian Schaller <uraeus@gnome.org>
33062 * gst-plugins-base.spec.in:
33063 update with latest files
33064 Original commit message from CVS:
33065 update with latest files
33067 2006-01-30 23:42:54 +0000 Sébastien Moutte <sebastien@moutte.net>
33069 win32/vs7: add vs7 project files created by Sergey Scobich
33070 Original commit message from CVS:
33072 add vs7 project files created by Sergey Scobich
33074 2006-01-30 22:18:53 +0000 Sébastien Moutte <sebastien@moutte.net>
33076 win32/vs8: add vs8 project files created by Sergey Scobich
33077 Original commit message from CVS:
33079 add vs8 project files created by Sergey Scobich
33081 2006-01-30 19:22:22 +0000 Andy Wingo <wingo@pobox.com>
33083 ext/ogg/gstoggmux.c (gst_ogg_mux_dequeue_page): Compare timestamp + duration, not just timestamp -- ogg pages should ...
33084 Original commit message from CVS:
33085 2006-01-30 Andy Wingo <wingo@pobox.com>
33086 * ext/ogg/gstoggmux.c (gst_ogg_mux_dequeue_page): Compare
33087 timestamp + duration, not just timestamp -- ogg pages should be
33088 ordered by stop time. Necessary fix given the change in vorbis
33091 2006-01-30 19:21:07 +0000 Andy Wingo <wingo@pobox.com>
33094 * ext/theora/gsttheoraenc.h:
33095 * ext/theora/theoraenc.c:
33096 * tests/check/pipelines/theoraenc.c:
33097 ext/theora/theoraenc.c (theora_enc_sink_setcaps)
33098 Original commit message from CVS:
33099 2006-01-30 Andy Wingo <wingo@pobox.com>
33100 * ext/theora/theoraenc.c (theora_enc_sink_setcaps)
33101 (gst_theora_enc_init): Pull the granule shift out of the encoder.
33102 (granulepos_add): New function, handles the messiness of adjusting
33104 (theora_buffer_from_packet):
33105 (theora_enc_chain):
33106 (theora_enc_sink_event): Use granulepos_add, not +.
33107 * tests/check/pipelines/theoraenc.c
33108 (check_buffer_granulepos_from_starttime): Just check the frame
33109 count, not the actual granulepos -- we can't dictate to the
33110 encoder when it should be placing keyframes.
33112 2006-01-30 18:17:19 +0000 Thomas Vander Stichele <thomas@apestaart.org>
33114 ext/gnomevfs/gstgnomevfssrc.c: SERVICE_NOT_AVAILABLE happens for example when you're trying to play an http:// stream...
33115 Original commit message from CVS:
33116 * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_start):
33117 SERVICE_NOT_AVAILABLE happens for example when you're trying to
33118 play an http:// stream from a server that's not serving
33120 2006-01-30 17:08:11 +0000 Andy Wingo <wingo@pobox.com>
33122 tests/check/pipelines/: Totally remove the UINT64_CONSTANT macro, doesn't appear to be needed or available.
33123 Original commit message from CVS:
33124 2006-01-30 Andy Wingo <wingo@pobox.com>
33125 * tests/check/pipelines/vorbisenc.c (TIMESTAMP_OFFSET):
33126 * tests/check/pipelines/theoraenc.c (TIMESTAMP_OFFSET): Totally
33127 remove the UINT64_CONSTANT macro, doesn't appear to be needed or
33130 2006-01-30 17:01:54 +0000 Andy Wingo <wingo@pobox.com>
33132 ext/theora/: Same changes as were done to vorbisenc, although theoraenc was timestamping correctly. Added handling of...
33133 Original commit message from CVS:
33134 2006-01-30 Andy Wingo <wingo@pobox.com>
33135 * ext/theora/gsttheoraenc.h:
33136 * ext/theora/theoraenc.c: Same changes as were done to vorbisenc,
33137 although theoraenc was timestamping correctly. Added handling of
33138 streams that start with nonzero timestamps.
33139 * tests/check/Makefile.am:
33140 * tests/check/pipelines/theoraenc.c: New file, basically does same
33141 tests as vorbisenc.
33142 * tests/check/pipelines/vorbisenc.c: I claim these bugs.
33144 2006-01-30 16:19:33 +0000 Wim Taymans <wim.taymans@gmail.com>
33146 gst-libs/gst/audio/gstaudiosink.c: Implement pause that does not wait for completion.
33147 Original commit message from CVS:
33148 * gst-libs/gst/audio/gstaudiosink.c:
33149 (gst_audioringbuffer_class_init), (gst_audioringbuffer_release),
33150 (gst_audioringbuffer_pause):
33151 Implement pause that does not wait for completion.
33152 * gst-libs/gst/audio/gstbaseaudiosink.c:
33153 (gst_base_audio_sink_render), (gst_base_audio_sink_change_state):
33154 Don't drop buffers when going to PAUSED but perform preroll on
33155 remaining samples now that core base class supports this.
33156 * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_release),
33157 (gst_ring_buffer_pause_unlocked), (gst_ring_buffer_stop),
33158 (gst_ring_buffer_commit):
33159 Pause should not signal waiters.
33160 Implement return value of _commit correctly.
33162 2006-01-30 15:01:28 +0000 Andy Wingo <wingo@pobox.com>
33164 tests/check/Makefile.am (check_vorbis): Add pipelines/vorbisenc.
33165 Original commit message from CVS:
33166 2006-01-30 Andy Wingo <wingo@pobox.com>
33167 * tests/check/Makefile.am (check_vorbis): Add pipelines/vorbisenc.
33168 * ext/vorbis/vorbisenc.c (gst_vorbisenc_buffer_from_packet): Logic
33169 updated to timestamp from the first sample, not the last.
33170 (gst_vorbisenc_buffer_from_header_packet): New function, takes
33171 special care of granulepos and timestamp for header packets.
33172 (gst_vorbisenc_chain): Reflow, fix some leaks, and handle the case
33173 when the first buffer has a nonzero timestamp.
33174 * ext/vorbis/vorbisenc.h (GstVorbisEnc.granulepos_offset)
33175 (GstVorbisEnc.subgranule_offset): New members. Take care of the
33176 case when the first audio buffer we get has a nonzero timestamp.
33177 (GstVorbisEnc.next_ts): Renamed from prev_ts, because now we
33178 properly timestamp vorbis buffers with the time of the first
33179 sample, not the last.
33180 * ext/vorbis/vorbisenc.c (granulepos_to_clocktime): Renamed from
33181 vorbis_granule_time_copy -- now it takes the granule/subgranule
33182 offset into account.
33183 * tests/check/pipelines/vorbisenc.c: New test for correctness of
33184 timestamps, durations, and granulepos on buffers produced by
33187 2006-01-30 14:42:28 +0000 Eric Jonas <jonas@mit.edu>
33189 gst/ffmpegcolorspace/gstffmpegcodecmap.c: Patch from Eric Jonas to support conversions to/from UYVY (Fixes: #324626)
33190 Original commit message from CVS:
33191 * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
33192 (gst_ffmpeg_pixfmt_to_caps), (gst_ffmpeg_caps_to_pixfmt):
33193 Patch from Eric Jonas to support conversions to/from UYVY
33196 2006-01-30 08:11:14 +0000 Julien Moutte <julien@moutte.net>
33198 gst/playback/: Implement subtitles.
33199 Original commit message from CVS:
33200 2006-01-30 Julien MOUTTE <julien@moutte.net>
33201 * gst/playback/gstplaybasebin.c: (group_commit),
33203 (setup_subtitle), (setup_source), (set_active_source):
33204 * gst/playback/gstplaybin.c: (gst_play_bin_dispose),
33205 (gen_text_element), (gen_audio_element), (gen_vis_element),
33206 (remove_sinks), (add_sink), (setup_sinks): Implement subtitles.
33208 2006-01-29 19:13:39 +0000 Sébastien Moutte <sebastien@moutte.net>
33210 gst-libs/gst/audio/audio.h: (GST_CLOCK_TIME_TO_FRAMES)
33211 Original commit message from CVS:
33212 * gst-libs/gst/audio/audio.h: (GST_CLOCK_TIME_TO_FRAMES)
33213 * gst-libs/gst/audio/gstbaseaudiosink.c: (gst_base_audio_sink_render)
33214 use of gst_guint64_to_gdouble to be compliant with vs6
33215 * gst/playback/gstdecodebin.c: (try_to_link_1)
33216 * gst/videorate/videorate.c: (gst_video_rate_blank_data)
33217 use of G_GINT64_CONSTANT for int64 constants
33218 * win32/common/libgstinterfaces.def:
33219 export some symbols (gst_mixer_get_type,gst_mixer_track_get_type)
33221 update and add new project files
33223 2006-01-29 18:21:12 +0000 Thomas Vander Stichele <thomas@apestaart.org>
33225 add a win32-update rule like in core, and copy over enumtypes files
33226 Original commit message from CVS:
33229 * win32/common/interfaces-enumtypes.c:
33230 (gst_color_balance_type_get_type), (gst_mixer_type_get_type),
33231 (gst_mixer_track_flags_get_type),
33232 (gst_tuner_channel_flags_get_type):
33233 * win32/common/interfaces-enumtypes.h:
33234 * win32/common/multichannel-enumtypes.c:
33235 (gst_audio_channel_position_get_type):
33236 * win32/common/multichannel-enumtypes.h:
33237 add a win32-update rule like in core, and copy over enumtypes files
33239 2006-01-29 18:07:51 +0000 Thomas Vander Stichele <thomas@apestaart.org>
33242 generate win32/common/config.h
33243 Original commit message from CVS:
33244 generate win32/common/config.h
33246 2006-01-29 18:05:42 +0000 Thomas Vander Stichele <thomas@apestaart.org>
33248 win32/: add config files just like in core
33249 Original commit message from CVS:
33251 * win32/common/config.h:
33252 * win32/common/config.h.in:
33253 add config files just like in core
33255 2006-01-28 18:22:06 +0000 Thomas Vander Stichele <thomas@apestaart.org>
33257 ext/alsa/: Update all error messages. All of them should either use the default translated message, or actually prov...
33258 Original commit message from CVS:
33259 * ext/alsa/gstalsasink.c: (gst_alsasink_init), (set_hwparams),
33260 (set_swparams), (gst_alsasink_prepare), (gst_alsasink_unprepare),
33261 (gst_alsasink_close), (gst_alsasink_write), (gst_alsasink_reset):
33262 * ext/alsa/gstalsasrc.c: (gst_alsasrc_init), (set_hwparams),
33263 (set_swparams), (gst_alsasrc_open), (gst_alsasrc_prepare),
33264 (gst_alsasrc_unprepare), (gst_alsasrc_read):
33265 Update all error messages. All of them should either use
33266 the default translated message, or actually provide a
33267 translatable string.
33268 Make the string for channel count problems meaningful.
33270 2006-01-28 18:19:18 +0000 Tim-Philipp Müller <tim@centricular.net>
33272 gst-libs/gst/audio/gstringbuffer.c: Make gcc-4.1 happy (part of #327357).
33273 Original commit message from CVS:
33274 * gst-libs/gst/audio/gstringbuffer.c: (build_linear_format):
33275 Make gcc-4.1 happy (part of #327357).
33277 2006-01-28 16:35:47 +0000 Thomas Vander Stichele <thomas@apestaart.org>
33279 sys/v4l/v4l_calls.c: check for and throw RESOURCE_BUSY
33280 Original commit message from CVS:
33281 * sys/v4l/v4l_calls.c: (gst_v4l_open):
33282 check for and throw RESOURCE_BUSY
33284 2006-01-28 02:13:14 +0000 David Schleef <ds@schleef.org>
33286 gst/videoscale/vs_scanline.c: Oops, *that's* why I never checked in this change -- it requires liboil features not in...
33287 Original commit message from CVS:
33288 * gst/videoscale/vs_scanline.c: Oops, *that's* why I never
33289 checked in this change -- it requires liboil features not
33290 in 0.3.6. Revert parts.
33292 2006-01-27 23:40:19 +0000 David Schleef <ds@schleef.org>
33294 update liboil requirement to 0.3.6
33295 Original commit message from CVS:
33297 * configure.ac: update liboil requirement to 0.3.6
33298 * gst/videoscale/Makefile.am:
33299 * gst/videoscale/vs_scanline.c: liboilify
33301 2006-01-27 17:00:09 +0000 Jan Schmidt <thaytan@mad.scientist.com>
33303 ext/libvisual/visual.c: When pad_alloc returns a GstFlowReturn other than GST_FLOW_OK, make sure it is passed upstream.
33304 Original commit message from CVS:
33305 * ext/libvisual/visual.c: (get_buffer):
33306 When pad_alloc returns a GstFlowReturn other
33307 than GST_FLOW_OK, make sure it is passed upstream.
33309 2006-01-27 01:36:01 +0000 Jan Schmidt <thaytan@mad.scientist.com>
33311 ext/alsa/gstalsasink.c: Free the device name string.
33312 Original commit message from CVS:
33313 * ext/alsa/gstalsasink.c: (gst_alsasink_finalise),
33314 (gst_alsasink_class_init):
33315 Free the device name string.
33316 * ext/ogg/gstoggmux.c: (gst_ogg_mux_class_init),
33317 (gst_ogg_mux_request_new_pad), (gst_ogg_mux_release_pad),
33318 (gst_ogg_mux_handle_src_event), (gst_ogg_mux_clear_collectpads):
33319 Don't remove a pad from the collectpads structure until it
33320 is released - it's a request pad, and may receive data again
33321 if the element gets moved back to PLAYING state.
33322 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support):
33323 Ensure we turn on double buffering on the Xv port, and
33324 set the colour key to something dark and mysterious that
33327 2006-01-27 01:06:29 +0000 Thomas Vander Stichele <thomas@apestaart.org>
33329 ext/: - a library should not call setlocale. see Libraries node in gettext manual
33330 Original commit message from CVS:
33331 * ext/alsa/gstalsaplugin.c: (plugin_init):
33332 * ext/cdparanoia/gstcdparanoiasrc.c:
33333 (gst_cd_paranoia_src_base_init), (plugin_init):
33334 * ext/gnomevfs/gstgnomevfs.c: (plugin_init):
33335 * ext/ogg/gstoggdemux.c: (gst_ogg_demux_plugin_init):
33336 - a library should not call setlocale. see Libraries node in
33338 - make sure all plugins that use translation do bindtextdomain
33339 to point to the localedir
33340 * gst/playback/gstplaybin.c: (gen_vis_element), (add_sink),
33341 (setup_sinks), (plugin_init):
33342 all this, and check for NULL when creating sinks
33344 2006-01-26 23:21:31 +0000 Julien Moutte <julien@moutte.net>
33346 gst/subparse/gstsubparse.c: Make typefinding of subtitles work again.
33347 Original commit message from CVS:
33348 2006-01-27 Julien MOUTTE <julien@moutte.net>
33349 * gst/subparse/gstsubparse.c: (gst_subparse_type_find),
33350 (plugin_init): Make typefinding of subtitles work again.
33352 2006-01-26 20:40:20 +0000 Tim-Philipp Müller <tim@centricular.net>
33354 gst/typefind/gsttypefindfunctions.c: Backport a bunch of typefinding fixes from the 0.8 branch.
33355 Original commit message from CVS:
33356 * gst/typefind/gsttypefindfunctions.c: (aac_type_find),
33357 (mp3_type_frame_length_from_header), (mp3_type_find),
33358 (wavpack_type_find), (m4a_type_find), (ircam_type_find),
33360 Backport a bunch of typefinding fixes from the 0.8 branch.
33361 Also, improve wavpack typefinding: if we can't peek the
33362 entire wavpack block, try to parse the bits we can get and
33363 see if we find what we're looking for in those.
33365 2006-01-26 19:17:38 +0000 Julien Moutte <julien@moutte.net>
33367 sys/: Handle some more cases of pixel aspect ratio.
33368 Original commit message from CVS:
33369 2006-01-26 Julien MOUTTE <julien@moutte.net>
33370 * sys/ximage/ximagesink.c:
33371 (gst_ximagesink_calculate_pixel_aspect_ratio):
33372 * sys/xvimage/xvimagesink.c:
33373 (gst_xvimagesink_calculate_pixel_aspect_ratio): Handle some
33374 more cases of pixel aspect ratio.
33376 2006-01-26 13:09:24 +0000 Edward Hervey <bilboed@bilboed.com>
33378 gst/playback/gstdecodebin.c: Also consider the flush-start and tag events as unblockers for the pad probes.
33379 Original commit message from CVS:
33380 * gst/playback/gstdecodebin.c: (pad_probe):
33381 Also consider the flush-start and tag events as unblockers
33382 for the pad probes.
33384 2006-01-26 12:32:58 +0000 Julien Moutte <julien@moutte.net>
33386 gst/playback/gstplaybin.c: On the fly visualisation switch, works disabling, enabling as well but it won't be able to...
33387 Original commit message from CVS:
33388 2006-01-26 Julien MOUTTE <julien@moutte.net>
33389 * gst/playback/gstplaybin.c: (gst_play_bin_init),
33390 (gst_play_bin_dispose), (gst_play_bin_vis_unblocked),
33391 (gst_play_bin_vis_blocked), (gst_play_bin_set_property):
33392 On the fly visualisation switch, works disabling, enabling as
33393 well but it won't be able to enable vis in a playbin that was
33394 created with no visualisation.
33396 2006-01-25 10:50:32 +0000 Wim Taymans <wim.taymans@gmail.com>
33398 gst-libs/gst/audio/gstbaseaudiosink.c: Undo previous commit, it breaks resume after pause.
33399 Original commit message from CVS:
33400 * gst-libs/gst/audio/gstbaseaudiosink.c:
33401 (gst_base_audio_sink_render), (gst_base_audio_sink_change_state):
33402 Undo previous commit, it breaks resume after pause.
33404 2006-01-25 09:27:01 +0000 Wim Taymans <wim.taymans@gmail.com>
33406 gst-libs/gst/audio/gstbaseaudiosink.c: Improve debugging.
33407 Original commit message from CVS:
33408 * gst-libs/gst/audio/gstbaseaudiosink.c:
33409 (gst_base_audio_sink_setcaps), (gst_base_audio_sink_event),
33410 (gst_base_audio_sink_preroll), (gst_base_audio_sink_render):
33412 Post error when caps cannot be parsed.
33413 Resync on discontinuity in the stream.
33414 Clip samples to segment boundaries.
33415 return WRONG_STATE sooner when we are flushing.
33416 * gst-libs/gst/audio/gstbaseaudiosrc.c: (gst_base_audio_src_init),
33417 (gst_base_audio_src_get_time), (gst_base_audio_src_create):
33418 Make audiosrc operate in TIME.
33419 Set TIMESTAMP and DURATION on buffers.
33421 2006-01-24 21:55:21 +0000 Tim-Philipp Müller <tim@centricular.net>
33423 tests/examples/seek/seek.c: Output tag messages as well.
33424 Original commit message from CVS:
33425 * tests/examples/seek/seek.c: (main):
33426 Output tag messages as well.
33428 2006-01-23 15:05:24 +0000 Edward Hervey <bilboed@bilboed.com>
33430 gst/playback/gstdecodebin.c: Replace GstPadBlockCallback with pad probes that detect first buffer AND eos before remo...
33431 Original commit message from CVS:
33432 * gst/playback/gstdecodebin.c: (gst_decode_bin_init),
33433 (free_pad_probes), (remove_fakesink), (pad_probe),
33434 (close_pad_link), (gst_decode_bin_change_state):
33435 Replace GstPadBlockCallback with pad probes that detect
33436 first buffer AND eos before removing fakesink.
33437 Fixes hang with demuxers doing EOS while pre-rolling.
33440 2006-01-23 10:10:36 +0000 Jens Granseuer <jensgr@gmx.net>
33442 GCC 2.95 fixes (#328263).
33443 Original commit message from CVS:
33444 2006-01-23 Andy Wingo <wingo@pobox.com>
33445 * ext/alsa/gstalsasink.c:
33446 * gst-libs/gst/rtp/gstbasertpdepayload.c:
33447 (gst_base_rtp_depayload_setcaps),
33448 (gst_base_rtp_depayload_add_to_queue),
33449 (gst_base_rtp_depayload_queue_release): GCC 2.95 fixes (#328263).
33450 Patch by: Jens Granseuer <jensgr at gmx dot net>
33452 2006-01-22 17:24:02 +0000 Julien Moutte <julien@moutte.net>
33454 sys/: Playbin keeps some ref to some frames. We might get a frame destroyed after changing state to
33455 Original commit message from CVS:
33456 2006-01-22 Julien MOUTTE <julien@moutte.net>
33457 * sys/ximage/ximagesink.c: (gst_ximagesink_ximage_destroy):
33458 * sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_destroy),
33459 (gst_xvimagesink_buffer_alloc): Playbin keeps some ref to some
33460 frames. We might get a frame destroyed after changing state to
33461 NULL, adding a safety check on xcontext.
33463 2006-01-22 14:50:53 +0000 Tim-Philipp Müller <tim@centricular.net>
33465 gst-libs/gst/interfaces/xoverlay.c: Fix prepare-xwindow-id code example in the docs - we need to ignore all messages ...
33466 Original commit message from CVS:
33467 * gst-libs/gst/interfaces/xoverlay.c:
33468 Fix prepare-xwindow-id code example in the docs - we need to
33469 ignore all messages that aren't element messages as well.
33471 2006-01-21 22:40:03 +0000 Julien Moutte <julien@moutte.net>
33473 sys/xvimage/xvimagesink.c: I think one day i'll completely undestand how caps negotiation is supposed to work. This r...
33474 Original commit message from CVS:
33475 2006-01-21 Julien MOUTTE <julien@moutte.net>
33476 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_buffer_alloc):
33477 I think one day i'll completely undestand how caps negotiation
33478 is supposed to work. This refactoring handles buffer_alloc
33479 called with caps we can't handle. We definitely don't want a
33480 set_caps with those caps, so we define and allocate a buffer
33481 we would like to receive.
33483 2006-01-20 19:10:26 +0000 Christian Schaller <uraeus@gnome.org>
33487 up automake requirement to 1.7
33488 Original commit message from CVS:
33489 up automake requirement to 1.7
33491 2006-01-19 10:59:51 +0000 Tim-Philipp Müller <tim@centricular.net>
33493 gst/playback/gstplaybasebin.c: Free iterator when done.
33494 Original commit message from CVS:
33495 * gst/playback/gstplaybasebin.c: (setup_source):
33496 Free iterator when done.
33498 2006-01-17 11:43:49 +0000 Jan Schmidt <thaytan@mad.scientist.com>
33500 gst-libs/gst/audio/gstbaseaudiosink.c: Fix playback of non-synchronised streams by assuming a rate of 1.0 instead of ...
33501 Original commit message from CVS:
33502 * gst-libs/gst/audio/gstbaseaudiosink.c:
33503 (gst_base_audio_sink_render):
33504 Fix playback of non-synchronised streams by assuming a rate
33505 of 1.0 instead of a random one.
33506 Makes this work again:
33507 gst-launch filesrc location=raw_audio.file ! 'audio/x-raw-int,
33508 endianness=(int)4321, signed=(boolean)true, width=(int)16,
33509 depth=(int)16, rate=(int)44100, channels=(int)2' ! audioconvert !
33510 audioresample ! alsasink
33512 2006-01-16 21:01:29 +0000 Thomas Vander Stichele <thomas@apestaart.org>
33516 Original commit message from CVS:
33519 === release 0.10.2 ===
33521 2006-01-16 20:59:32 +0000 Thomas Vander Stichele <thomas@apestaart.org>
33527 * docs/plugins/gst-plugins-base-plugins.args:
33528 * docs/plugins/inspect/plugin-adder.xml:
33529 * docs/plugins/inspect/plugin-alsa.xml:
33530 * docs/plugins/inspect/plugin-audioconvert.xml:
33531 * docs/plugins/inspect/plugin-audiorate.xml:
33532 * docs/plugins/inspect/plugin-audioresample.xml:
33533 * docs/plugins/inspect/plugin-audiotestsrc.xml:
33534 * docs/plugins/inspect/plugin-cdparanoia.xml:
33535 * docs/plugins/inspect/plugin-decodebin.xml:
33536 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
33537 * docs/plugins/inspect/plugin-gnomevfs.xml:
33538 * docs/plugins/inspect/plugin-libvisual.xml:
33539 * docs/plugins/inspect/plugin-ogg.xml:
33540 * docs/plugins/inspect/plugin-pango.xml:
33541 * docs/plugins/inspect/plugin-playbin.xml:
33542 * docs/plugins/inspect/plugin-subparse.xml:
33543 * docs/plugins/inspect/plugin-tcp.xml:
33544 * docs/plugins/inspect/plugin-theora.xml:
33545 * docs/plugins/inspect/plugin-typefindfunctions.xml:
33546 * docs/plugins/inspect/plugin-video4linux.xml:
33547 * docs/plugins/inspect/plugin-videorate.xml:
33548 * docs/plugins/inspect/plugin-videoscale.xml:
33549 * docs/plugins/inspect/plugin-videotestsrc.xml:
33550 * docs/plugins/inspect/plugin-volume.xml:
33551 * docs/plugins/inspect/plugin-vorbis.xml:
33552 * docs/plugins/inspect/plugin-ximagesink.xml:
33553 * docs/plugins/inspect/plugin-xvimagesink.xml:
33555 Original commit message from CVS:
33558 2006-01-16 16:38:15 +0000 Jan Schmidt <thaytan@mad.scientist.com>
33560 gst/playback/: Comment out broken code that connects to the state-changed signal.
33561 Original commit message from CVS:
33562 * gst/playback/gststreaminfo.c: (gst_stream_info_set_mute):
33563 * gst/playback/gststreamselector.c:
33564 (gst_stream_selector_set_property):
33565 Comment out broken code that connects to the state-changed signal.
33566 At this point, changing current stream selection is broken, but
33567 stuff like gst-launch playbin current-audio=1 works and filters
33568 to the chosen stream.
33570 2006-01-16 15:31:14 +0000 Thomas Vander Stichele <thomas@apestaart.org>
33572 ext/vorbis/vorbisdec.c: Fix #327216 (null dereference in vorbisdec)
33573 Original commit message from CVS:
33574 * ext/vorbis/vorbisdec.c: (vorbis_dec_src_query):
33575 Fix #327216 (null dereference in vorbisdec)
33577 2006-01-16 15:19:55 +0000 Tim-Philipp Müller <tim@centricular.net>
33579 ext/theora/theoradec.c: Post taglist actually on bus instead of just freeing it (fixes #327114 and totem bug #327080).
33580 Original commit message from CVS:
33581 * ext/theora/theoradec.c: (theora_handle_comment_packet):
33582 Post taglist actually on bus instead of just freeing it
33583 (fixes #327114 and totem bug #327080).
33584 * ext/vorbis/vorbisdec.c: (vorbis_handle_comment_packet):
33585 Use gst_element_found_tags_for_pad(), so that the tags
33586 are sent downstream as an event as well.
33588 2006-01-15 10:06:40 +0000 Thomas Vander Stichele <thomas@apestaart.org>
33590 sys/: move all regularly occurring messages to GST_LOG level add some more object logs
33591 Original commit message from CVS:
33592 * sys/ximage/ximagesink.c: (gst_ximage_buffer_finalize),
33593 (gst_ximagesink_ximage_new), (gst_ximagesink_ximage_put),
33594 (gst_ximagesink_buffer_alloc):
33595 * sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_finalize),
33596 (gst_xvimagesink_xvimage_put), (gst_xvimagesink_show_frame),
33597 (gst_xvimagesink_buffer_alloc):
33598 move all regularly occurring messages to GST_LOG level
33599 add some more object logs
33601 2006-01-14 22:59:49 +0000 Thomas Vander Stichele <thomas@apestaart.org>
33619 Original commit message from CVS:
33622 2006-01-14 20:46:25 +0000 Thomas Vander Stichele <thomas@apestaart.org>
33624 ext/ogg/gstoggmux.c: fix a silly segfault
33625 Original commit message from CVS:
33626 2006-01-14 Thomas Vander Stichele <thomas at apestaart dot org>
33627 * ext/ogg/gstoggmux.c: (gst_ogg_mux_collected):
33628 fix a silly segfault
33630 2006-01-14 12:52:22 +0000 Tim-Philipp Müller <tim@centricular.net>
33632 Add docs for mixerutils stuff.
33633 Original commit message from CVS:
33634 * docs/libs/gst-plugins-base-libs-docs.sgml:
33635 * docs/libs/gst-plugins-base-libs-sections.txt:
33636 * gst-libs/gst/audio/mixerutils.c:
33637 * gst-libs/gst/audio/mixerutils.h:
33638 Add docs for mixerutils stuff.
33640 2006-01-13 17:17:07 +0000 Tim-Philipp Müller <tim@centricular.net>
33642 gst/playback/gstplaybasebin.c: Fix playback for sources that emit raw audio or raw video streams (e.g.: cd audio sour...
33643 Original commit message from CVS:
33644 * gst/playback/gstplaybasebin.c: (setup_source):
33645 Fix playback for sources that emit raw audio or
33646 raw video streams (e.g.: cd audio sources) (#325984).
33648 2006-01-13 16:45:50 +0000 Thomas Vander Stichele <thomas@apestaart.org>
33650 gst-libs/gst/audio/mixerutils.c: actually save the element we create
33651 Original commit message from CVS:
33652 * gst-libs/gst/audio/mixerutils.c:
33653 (gst_audio_mixer_filter_do_filter):
33654 actually save the element we create
33656 2006-01-13 16:17:50 +0000 Christian Schaller <uraeus@gnome.org>
33658 * gst-plugins-base.spec.in:
33659 remove version suffix
33660 Original commit message from CVS:
33661 remove version suffix
33663 2006-01-12 14:56:11 +0000 Tim-Philipp Müller <tim@centricular.net>
33665 gst-libs/gst/cdda/gstcddabasesrc.c: No need to post a tag message on the bus when seeking within the same track, only...
33666 Original commit message from CVS:
33667 * gst-libs/gst/cdda/gstcddabasesrc.c:
33668 (gst_cdda_base_src_handle_track_seek):
33669 No need to post a tag message on the bus when seeking
33670 within the same track, only post it when the current
33673 2006-01-11 18:30:25 +0000 Jan Schmidt <thaytan@mad.scientist.com>
33675 gst/playback/: Reenable stream selection. These mechanisms need a complete overhaul in the face of 0.8->0.10 changes ...
33676 Original commit message from CVS:
33677 * gst/playback/gstplaybasebin.c: (group_destroy),
33678 (probe_triggered), (new_decoded_pad), (mute_group_type),
33679 (set_active_source):
33680 * gst/playback/gststreaminfo.c: (gst_stream_info_set_mute):
33681 * gst/playback/gststreamselector.c:
33682 (gst_stream_selector_base_init),
33683 (gst_stream_selector_set_property),
33684 (gst_stream_selector_request_new_pad):
33685 Reenable stream selection. These mechanisms need a complete overhaul
33686 in the face of 0.8->0.10 changes though.
33688 2006-01-11 18:03:24 +0000 Jan Schmidt <thaytan@mad.scientist.com>
33690 ext/ogg/gstoggdemux.c: Change the pad template to src_%d to match the pads that are created from it. decodebin needs ...
33691 Original commit message from CVS:
33692 * ext/ogg/gstoggdemux.c:
33693 Change the pad template to src_%d to match the pads that
33694 are created from it. decodebin needs this information in order
33695 to decide that oggdemux is capable of producing multiple pads
33696 (and hence needs queues inserted).
33697 * ext/ogg/gstoggmux.c: (gst_ogg_mux_queue_pads),
33698 (gst_ogg_mux_collected):
33699 Make debug output more useful by using GST_PTR_FORMAT.
33701 2006-01-11 17:38:35 +0000 Christian Schaller <uraeus@gnome.org>
33703 * gst-plugins-base.spec.in:
33704 update spec.in file
33705 Original commit message from CVS:
33706 update spec.in file
33708 2006-01-11 15:11:20 +0000 Tim-Philipp Müller <tim@centricular.net>
33710 gst-libs/gst/audio/gstringbuffer.c: Set depth and width for alaw/mulaw (fixes #326601).
33711 Original commit message from CVS:
33712 Reviewed by: Tim-Philipp Müller <tim at centricular dot net>
33713 * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_parse_caps):
33714 Set depth and width for alaw/mulaw (fixes #326601).
33716 2006-01-10 23:58:36 +0000 Thomas Vander Stichele <thomas@apestaart.org>
33718 tests/icles/Makefile.am: don't build the tests if we don't have the libs
33719 Original commit message from CVS:
33720 * tests/icles/Makefile.am:
33721 don't build the tests if we don't have the libs
33723 2006-01-10 18:06:56 +0000 Tim-Philipp Müller <tim@centricular.net>
33725 ext/cdparanoia/gstcdparanoiasrc.c: Don't try to free NULL pointers.
33726 Original commit message from CVS:
33727 * ext/cdparanoia/gstcdparanoiasrc.c: (gst_cd_paranoia_src_close),
33728 (gst_cd_paranoia_paranoia_callback):
33729 Don't try to free NULL pointers.
33731 2006-01-10 15:47:48 +0000 Edward Hervey <bilboed@bilboed.com>
33733 gst/audiorate/gstaudiorate.c: Add debugging category.
33734 Original commit message from CVS:
33735 * gst/audiorate/gstaudiorate.c: (gst_audio_rate_chain),
33736 (gst_audio_rate_change_state), (plugin_init):
33737 Add debugging category.
33739 Add case for incoming buffers without valid offset/offset_end.
33741 2006-01-10 12:25:59 +0000 Michael Smith <msmith@xiph.org>
33743 gst-libs/gst/audio/gstaudiosrc.c: Don't leak GCond in audio sources.
33744 Original commit message from CVS:
33745 * gst-libs/gst/audio/gstaudiosrc.c: (gst_audioringbuffer_dispose):
33746 Don't leak GCond in audio sources.
33748 2006-01-10 11:49:28 +0000 Jan Schmidt <thaytan@mad.scientist.com>
33750 gst/playback/gstplaybin.c: Don't leak an autoaudiosink/alsasink when we generate a new audio element. (old code, I gu...
33751 Original commit message from CVS:
33752 * gst/playback/gstplaybin.c: (gen_audio_element):
33753 Don't leak an autoaudiosink/alsasink when we generate
33754 a new audio element. (old code, I guess)
33756 2006-01-10 11:04:21 +0000 Michael Smith <msmith@xiph.org>
33758 gst/audiorate/gstaudiorate.c: Support float audio in audiorate.
33759 Original commit message from CVS:
33760 * gst/audiorate/gstaudiorate.c: (gst_audio_rate_setcaps):
33761 Support float audio in audiorate.
33762 Use width rather than depth for selecting sample width.
33764 2006-01-10 10:06:53 +0000 Tim-Philipp Müller <tim@centricular.net>
33766 gst/videotestsrc/videotestsrc.h: Use GLib types here (that way we don't have to include the generated _stdint.h heade...
33767 Original commit message from CVS:
33768 * gst/videotestsrc/videotestsrc.h:
33769 Use GLib types here (that way we don't have to include the
33770 generated _stdint.h header, which makes life easier for win32
33771 folks that don't use autotools for the build) (#325990, patch
33772 by: Sergey Scobich).
33774 2006-01-10 09:38:44 +0000 Tim-Philipp Müller <tim@centricular.net>
33776 gst-libs/gst/audio/gstringbuffer.*: Name (private) union, makes Forte compiler happy (this time for real) (#324900).
33777 Original commit message from CVS:
33778 * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_init),
33779 (gst_ring_buffer_set_flushing), (gst_ring_buffer_start),
33780 (gst_ring_buffer_pause), (wait_segment):
33781 * gst-libs/gst/audio/gstringbuffer.h:
33782 Name (private) union, makes Forte compiler happy (this time
33783 for real) (#324900).
33785 2006-01-09 10:52:33 +0000 Tim-Philipp Müller <tim@centricular.net>
33787 gst-libs/gst/audio/Makefile.am: Link against libgstinterfaces, needed for mixer and property probe stuff.
33788 Original commit message from CVS:
33789 * gst-libs/gst/audio/Makefile.am:
33790 Link against libgstinterfaces, needed for mixer
33791 and property probe stuff.
33793 2006-01-09 10:46:52 +0000 Edward Hervey <bilboed@bilboed.com>
33795 gst-libs/gst/Makefile.am:
33796 Original commit message from CVS:
33797 * gst-libs/gst/Makefile.am:
33799 2006-01-09 09:38:34 +0000 Tim-Philipp Müller <tim@centricular.net>
33801 gst-libs/gst/audio/: Add gst_audio_default_registry_mixer_filter() utility function.
33802 Original commit message from CVS:
33803 * gst-libs/gst/audio/Makefile.am:
33804 * gst-libs/gst/audio/mixerutils.c:
33805 (gst_audio_mixer_filter_do_filter),
33806 (gst_audio_mixer_filter_check_element),
33807 (gst_audio_mixer_filter_probe_feature),
33808 (element_factory_rank_compare_func),
33809 (gst_audio_default_registry_mixer_filter):
33810 * gst-libs/gst/audio/mixerutils.h:
33811 Add gst_audio_default_registry_mixer_filter() utility
33814 2006-01-03 17:33:16 +0000 Michael Smith <msmith@xiph.org>
33816 gst/audioresample/resample.h: As before, but for o_buf
33817 Original commit message from CVS:
33818 * gst/audioresample/resample.h:
33819 As before, but for o_buf
33821 2006-01-03 17:27:13 +0000 Michael Smith <msmith@xiph.org>
33823 gst/audioresample/resample.h: Declare struct _ResampleState.buffer as unsigned char *, not void *, since we do arithm...
33824 Original commit message from CVS:
33825 * gst/audioresample/resample.h:
33826 Declare struct _ResampleState.buffer as unsigned char *, not void *,
33827 since we do arithmetic on it.
33829 2006-01-02 23:37:38 +0000 Tim-Philipp Müller <tim@centricular.net>
33831 gst-libs/gst/audio/gstringbuffer.*: Sun's Forte compiler doesn't seem to like anonymous structs, so use same setup as...
33832 Original commit message from CVS:
33833 * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_init),
33834 (gst_ring_buffer_set_flushing), (gst_ring_buffer_start),
33835 (gst_ring_buffer_pause), (wait_segment):
33836 * gst-libs/gst/audio/gstringbuffer.h:
33837 Sun's Forte compiler doesn't seem to like anonymous structs,
33838 so use same setup as in GstBaseSrc (fixes #324900).
33840 2005-12-30 14:54:06 +0000 Stefan Kost <ensonic@users.sourceforge.net>
33842 move old example to tests/examples/volume/volune.c
33843 Original commit message from CVS:
33845 * gst/volume/Makefile.am:
33846 * gst/volume/demo.c:
33847 move old example to tests/examples/volume/volune.c
33848 * tests/examples/Makefile.am:
33849 * tests/examples/seek/seek.c: (main):
33850 change window-close event from "delete-event" to "destroy"
33851 * tests/examples/volume/Makefile.am:
33852 * tests/examples/volume/volume.c: (value_changed_callback),
33853 (setup_gui), (message_received), (eos_message_received), (main):
33854 fix event handling and bus usage
33856 2005-12-29 20:37:23 +0000 Stefan Kost <ensonic@users.sourceforge.net>
33858 gst/audiotestsrc/gstaudiotestsrc.*: update to basesrc changes, implement segmented seeking and eos handling, add a 's...
33859 Original commit message from CVS:
33860 * gst/audiotestsrc/gstaudiotestsrc.c:
33861 (gst_audiostestsrc_wave_get_type), (gst_audio_test_src_class_init),
33862 (gst_audio_test_src_init), (gst_audio_test_src_src_fixate),
33863 (gst_audio_test_src_query), (gst_audio_test_src_create_sine),
33864 (gst_audio_test_src_create_square),
33865 (gst_audio_test_src_create_saw),
33866 (gst_audio_test_src_create_triangle),
33867 (gst_audio_test_src_create_silence),
33868 (gst_audio_test_src_create_white_noise),
33869 (gst_audio_test_src_create_pink_noise),
33870 (gst_audio_test_src_init_sine_table),
33871 (gst_audio_test_src_create_sine_table),
33872 (gst_audio_test_src_change_wave),
33873 (gst_audio_test_src_change_volume), (gst_audio_test_src_do_seek),
33874 (gst_audio_test_src_create), (gst_audio_test_src_set_property):
33875 * gst/audiotestsrc/gstaudiotestsrc.h:
33876 update to basesrc changes, implement segmented seeking and eos handling,
33877 add a 'sine-tab' waveform for performance critical playback
33879 2005-12-29 16:17:55 +0000 Tim-Philipp Müller <tim@centricular.net>
33881 po/POTFILES.in: ... and this time the other modified file that I missed last time.
33882 Original commit message from CVS:
33884 ... and this time the other modified file that I missed last time.
33886 2005-12-29 14:31:49 +0000 Michael Smith <msmith@xiph.org>
33888 gst/playback/gstdecodebin.c: Fix non-C89 variable declaration not at the start of a block. Should help some compilers.
33889 Original commit message from CVS:
33890 * gst/playback/gstdecodebin.c: (new_pad):
33891 Fix non-C89 variable declaration not at the start of a block. Should
33892 help some compilers.
33894 2005-12-29 12:43:22 +0000 Tim-Philipp Müller <tim@centricular.net>
33896 tests/check/Makefile.am: And now fix 'make distcheck' (builddir != srcdir)
33897 Original commit message from CVS:
33898 * tests/check/Makefile.am:
33899 And now fix 'make distcheck' (builddir != srcdir)
33901 2005-12-29 12:22:24 +0000 Tim-Philipp Müller <tim@centricular.net>
33903 New cdparanoiasrc element based on cddabasesrc; enable cdparanoia plugin again (there are still fixes required to pla...
33904 Original commit message from CVS:
33906 * ext/cdparanoia/Makefile.am:
33907 * ext/cdparanoia/gstcdparanoia.c:
33908 * ext/cdparanoia/gstcdparanoia.h:
33909 * ext/cdparanoia/gstcdparanoiasrc.c:
33910 (gst_cd_paranoia_mode_get_type), (gst_cd_paranoia_src_base_init),
33911 (gst_cd_paranoia_src_init), (gst_cd_paranoia_src_class_init),
33912 (gst_cd_paranoia_src_open), (gst_cd_paranoia_src_close),
33913 (gst_cd_paranoia_paranoia_callback),
33914 (gst_cd_paranoia_src_read_sector), (gst_cd_paranoia_src_finalize),
33915 (gst_cd_paranoia_src_set_property),
33916 (gst_cd_paranoia_src_get_property), (plugin_init):
33917 * ext/cdparanoia/gstcdparanoiasrc.h:
33918 New cdparanoiasrc element based on cddabasesrc; enable cdparanoia
33919 plugin again (there are still fixes required to playbin to make
33920 cdda:// uris work there).
33922 2005-12-29 12:13:57 +0000 Tim-Philipp Müller <tim@centricular.net>
33924 tests/check/Makefile.am: Fix test case compilation.
33925 Original commit message from CVS:
33926 * tests/check/Makefile.am:
33927 Fix test case compilation.
33929 2005-12-29 11:49:11 +0000 Tim-Philipp Müller <tim@centricular.net>
33931 gst-libs/gst/cdda/gstcddabasesrc.c: An integer is not a string. Fix access to uninitialised variable.
33932 Original commit message from CVS:
33933 * gst-libs/gst/cdda/gstcddabasesrc.c:
33934 (gst_cdda_base_src_update_duration),
33935 (gst_cdda_base_src_calculate_cddb_id):
33936 An integer is not a string. Fix access to uninitialised variable.
33937 * tests/check/Makefile.am:
33938 Add cddabasesrc unit test; also actually enable the vorbis test.
33939 * tests/check/generic/states.c:
33940 Blacklist new cd audio elements as well.
33941 * tests/check/libs/cddabasesrc.c:
33942 Unit test for GstCddaBaseSrc (discid calculation mostly).
33944 2005-12-28 18:19:25 +0000 Tim-Philipp Müller <tim@centricular.net>
33946 docs/libs/: Add docs for libgstcdda/GstCddaBaseSrc.
33947 Original commit message from CVS:
33948 * docs/libs/Makefile.am:
33949 * docs/libs/gst-plugins-base-libs-docs.sgml:
33950 * docs/libs/gst-plugins-base-libs-sections.txt:
33951 * docs/libs/gst-plugins-base-libs.types:
33952 Add docs for libgstcdda/GstCddaBaseSrc.
33953 * gst-libs/gst/interfaces/mixertrack.h:
33954 Do one struct member per line with a semicolon at the end, that way
33955 even gtk-doc might parse it without complaining.
33957 2005-12-28 18:06:50 +0000 Tim-Philipp Müller <tim@centricular.net>
33959 Add new libgstcdda with GstCddaBaseSrc class.
33960 Original commit message from CVS:
33962 * gst-libs/gst/Makefile.am:
33963 * gst-libs/gst/cdda/Makefile.am:
33964 * gst-libs/gst/cdda/base64.c:
33965 * gst-libs/gst/cdda/base64.h:
33966 * gst-libs/gst/cdda/gstcddabasesrc.c:
33967 (gst_cdda_base_src_mode_get_type), (gst_cdda_base_src_base_init),
33968 (gst_cdda_base_src_class_init), (gst_cdda_base_src_init),
33969 (gst_cdda_base_src_finalize), (gst_cdda_base_src_set_property),
33970 (gst_cdda_base_src_get_property),
33971 (gst_cdda_base_src_get_track_from_sector),
33972 (gst_cdda_base_src_get_query_types), (gst_cdda_base_src_convert),
33973 (gst_cdda_base_src_query), (gst_cdda_base_src_is_seekable),
33974 (gst_cdda_base_src_do_seek), (gst_cdda_base_src_handle_track_seek),
33975 (gst_cdda_base_src_handle_event), (gst_cdda_base_src_uri_get_type),
33976 (gst_cdda_base_src_uri_get_protocols),
33977 (gst_cdda_base_src_uri_get_uri), (gst_cdda_base_src_uri_set_uri),
33978 (gst_cdda_base_src_uri_handler_init),
33979 (gst_cdda_base_src_setup_interfaces),
33980 (gst_cdda_base_src_add_track), (gst_cdda_base_src_update_duration),
33981 (cddb_sum), (gst_cddabasesrc_calculate_musicbrainz_discid),
33982 (lba_to_msf), (gst_cdda_base_src_calculate_cddb_id),
33983 (gst_cdda_base_src_add_tags),
33984 (gst_cdda_base_src_add_index_associations),
33985 (gst_cdda_base_src_set_index), (gst_cdda_base_src_get_index),
33986 (gst_cdda_base_src_track_sort_func), (gst_cdda_base_src_start),
33987 (gst_cdda_base_src_clear_tracks), (gst_cdda_base_src_stop),
33988 (gst_cdda_base_src_create):
33989 * gst-libs/gst/cdda/gstcddabasesrc.h:
33990 * gst-libs/gst/cdda/sha1.c:
33991 * gst-libs/gst/cdda/sha1.h:
33992 Add new libgstcdda with GstCddaBaseSrc class.
33994 2005-12-28 14:59:41 +0000 Tim-Philipp Müller <tim@centricular.net>
33996 ext/gnomevfs/gstgnomevfssink.h: Use GstBaseSinkClass as parent_class member for class struct, not
33997 Original commit message from CVS:
33998 * ext/gnomevfs/gstgnomevfssink.h:
33999 Use GstBaseSinkClass as parent_class member for class struct, not
34002 2005-12-27 22:29:43 +0000 Tim-Philipp Müller <tim@centricular.net>
34004 gst/videotestsrc/gstvideotestsrc.c: Add start method to reset running time and number of frames sent
34005 Original commit message from CVS:
34006 * gst/videotestsrc/gstvideotestsrc.c:
34007 (gst_video_test_src_class_init), (gst_video_test_src_start):
34008 Add start method to reset running time and number of frames sent
34009 when starting up (fixes #324696; patch by: Michal Benes).
34011 2005-12-27 21:58:28 +0000 Tim-Philipp Müller <tim@centricular.net>
34013 docs/plugins/: Add docs stuff for gnomevfssrc and gnomevfssink.
34014 Original commit message from CVS:
34015 * docs/plugins/Makefile.am:
34016 * docs/plugins/gst-plugins-base-plugins-docs.sgml:
34017 * docs/plugins/gst-plugins-base-plugins-sections.txt:
34018 * docs/plugins/gst-plugins-base-plugins.args:
34019 * docs/plugins/gst-plugins-base-plugins.hierarchy:
34020 * docs/plugins/gst-plugins-base-plugins.signals:
34021 Add docs stuff for gnomevfssrc and gnomevfssink.
34022 * ext/gnomevfs/gstgnomevfssrc.c:
34023 Fix example pipeline in gtk-doc blurb.
34025 2005-12-27 21:42:23 +0000 Tim-Philipp Müller <tim@centricular.net>
34027 ext/gnomevfs/: Port gnomevfssink; add gtk-doc blurb.
34028 Original commit message from CVS:
34029 * ext/gnomevfs/Makefile.am:
34030 * ext/gnomevfs/gstgnomevfs.c: (gst_gnome_vfs_uri_get_type),
34031 (gst_gnome_vfs_handle_copy), (gst_gnome_vfs_handle_free),
34032 (gst_gnome_vfs_handle_get_type), (plugin_init):
34033 * ext/gnomevfs/gstgnomevfs.h:
34034 * ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_do_init),
34035 (gst_gnome_vfs_sink_base_init), (gst_gnome_vfs_sink_class_init),
34036 (gst_gnome_vfs_sink_finalize), (gst_gnome_vfs_sink_init),
34037 (gst_gnome_vfs_sink_set_property),
34038 (gst_gnome_vfs_sink_get_property), (gst_gnome_vfs_sink_open_file),
34039 (gst_gnome_vfs_sink_close_file), (gst_gnome_vfs_sink_start),
34040 (gst_gnome_vfs_sink_stop), (gst_gnome_vfs_sink_handle_event),
34041 (gst_gnome_vfs_sink_query), (gst_gnome_vfs_sink_render),
34042 (gst_gnome_vfs_sink_uri_get_type),
34043 (gst_gnome_vfs_sink_uri_get_protocols),
34044 (gst_gnome_vfs_sink_uri_get_uri), (gst_gnome_vfs_sink_uri_set_uri),
34045 (gst_gnome_vfs_sink_uri_handler_init):
34046 * ext/gnomevfs/gstgnomevfssink.h:
34047 Port gnomevfssink; add gtk-doc blurb.
34048 * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_get_type),
34049 (gst_gnome_vfs_src_base_init), (gst_gnome_vfs_src_class_init),
34050 (gst_gnome_vfs_src_init), (gst_gnome_vfs_src_finalize),
34051 (gst_gnome_vfs_src_uri_get_type),
34052 (gst_gnome_vfs_src_uri_get_protocols),
34053 (gst_gnome_vfs_src_uri_get_uri), (gst_gnome_vfs_src_uri_set_uri),
34054 (gst_gnome_vfs_src_uri_handler_init),
34055 (gst_gnome_vfs_src_set_property), (gst_gnome_vfs_src_get_property),
34056 (gst_gnome_vfs_src_unicodify), (audiocast_thread_run),
34057 (gst_gnome_vfs_src_send_additional_headers_callback),
34058 (gst_gnome_vfs_src_received_headers_callback),
34059 (gst_gnome_vfs_src_push_callbacks),
34060 (gst_gnome_vfs_src_pop_callbacks),
34061 (gst_gnome_vfs_src_get_icy_metadata), (gst_gnome_vfs_src_create),
34062 (gst_gnome_vfs_src_is_seekable), (gst_gnome_vfs_src_get_size),
34063 (gst_gnome_vfs_src_start), (gst_gnome_vfs_src_stop):
34064 * ext/gnomevfs/gstgnomevfssrc.h:
34065 s/gst_gnomevfssrc/gst_gnome_vfs_src/; move header stuff to header
34066 file; add gtk-doc blurb with example pipelines.
34068 2005-12-23 18:16:22 +0000 Thomas Vander Stichele <thomas@apestaart.org>
34072 Original commit message from CVS:
34075 === release 0.10.1 ===
34077 2005-12-23 18:08:39 +0000 Thomas Vander Stichele <thomas@apestaart.org>
34083 * docs/libs/tmpl/gstcolorbalance.sgml:
34084 * docs/plugins/gst-plugins-base-plugins.args:
34085 * docs/plugins/gst-plugins-base-plugins.signals:
34086 * docs/plugins/inspect/plugin-adder.xml:
34087 * docs/plugins/inspect/plugin-alsa.xml:
34088 * docs/plugins/inspect/plugin-audioconvert.xml:
34089 * docs/plugins/inspect/plugin-audiorate.xml:
34090 * docs/plugins/inspect/plugin-audioresample.xml:
34091 * docs/plugins/inspect/plugin-audiotestsrc.xml:
34092 * docs/plugins/inspect/plugin-decodebin.xml:
34093 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
34094 * docs/plugins/inspect/plugin-gnomevfs.xml:
34095 * docs/plugins/inspect/plugin-libvisual.xml:
34096 * docs/plugins/inspect/plugin-ogg.xml:
34097 * docs/plugins/inspect/plugin-pango.xml:
34098 * docs/plugins/inspect/plugin-playbin.xml:
34099 * docs/plugins/inspect/plugin-subparse.xml:
34100 * docs/plugins/inspect/plugin-tcp.xml:
34101 * docs/plugins/inspect/plugin-theora.xml:
34102 * docs/plugins/inspect/plugin-typefindfunctions.xml:
34103 * docs/plugins/inspect/plugin-video4linux.xml:
34104 * docs/plugins/inspect/plugin-videorate.xml:
34105 * docs/plugins/inspect/plugin-videoscale.xml:
34106 * docs/plugins/inspect/plugin-videotestsrc.xml:
34107 * docs/plugins/inspect/plugin-volume.xml:
34108 * docs/plugins/inspect/plugin-vorbis.xml:
34109 * docs/plugins/inspect/plugin-ximagesink.xml:
34110 * docs/plugins/inspect/plugin-xvimagesink.xml:
34112 Original commit message from CVS:
34115 2005-12-21 20:59:52 +0000 Edgard Lima <edgard.lima@indt.org.br>
34118 * gst/typefind/gsttypefindfunctions.c:
34119 iLBC30 and iLBC20 added to typefind.
34120 Original commit message from CVS:
34121 iLBC30 and iLBC20 added to typefind.
34123 2005-12-20 15:57:06 +0000 Thomas Vander Stichele <thomas@apestaart.org>
34127 * docs/libs/tmpl/gstcolorbalance.sgml:
34143 Original commit message from CVS:
34146 2005-12-20 12:24:29 +0000 Thomas Vander Stichele <thomas@apestaart.org>
34148 * gst-libs/gst/audio/gstbaseaudiosink.c:
34149 * gst-libs/gst/audio/gstbaseaudiosrc.c:
34150 stop making fun of older compilers
34151 Original commit message from CVS:
34152 stop making fun of older compilers
34154 2005-12-20 12:00:26 +0000 Thomas Vander Stichele <thomas@apestaart.org>
34156 gst-libs/gst/audio/: update strings, values are in microseconds change the default sink buffer time to something that...
34157 Original commit message from CVS:
34158 * gst-libs/gst/audio/gstbaseaudiosink.c:
34159 (gst_base_audio_sink_class_init):
34160 * gst-libs/gst/audio/gstbaseaudiosrc.c:
34161 (gst_base_audio_src_class_init):
34162 update strings, values are in microseconds
34163 change the default sink buffer time to something that is smaller
34164 (to help software volume mixing have a slightly lower delay) but
34165 still be acceptable on Wim's laptop
34167 2005-12-20 10:13:05 +0000 Edward Hervey <bilboed@bilboed.com>
34169 gst-libs/gst/riff/riff-media.c: Made a quack, forgot to add DUCK to the riff video template.
34170 Original commit message from CVS:
34171 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_template_caps):
34172 Made a quack, forgot to add DUCK to the riff video template.
34174 2005-12-19 15:00:38 +0000 Edward Hervey <bilboed@bilboed.com>
34176 ext/ogg/gstogmparse.c: Make sure pads are initialized correctly.
34177 Original commit message from CVS:
34178 * ext/ogg/gstogmparse.c: (gst_ogm_text_parse_base_init),
34179 (gst_ogm_parse_init), (gst_ogm_audio_parse_init),
34180 (gst_ogm_video_parse_init), (gst_ogm_text_parse_init),
34181 (gst_ogm_parse_chain):
34182 Make sure pads are initialized correctly.
34183 * gst-libs/gst/riff/riff-ids.h:
34184 * gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
34185 (gst_riff_create_video_template_caps):
34186 Add a whole bunch of FOURCC <=> MimeType.
34187 Extend the riff video pad template to support the newly added fourcc.
34189 2005-12-18 15:04:21 +0000 Jan Schmidt <thaytan@mad.scientist.com>
34191 ext/ogg/gstoggdemux.c: Extra debug output when activating/deactivating chains.
34192 Original commit message from CVS:
34193 * ext/ogg/gstoggdemux.c: (gst_ogg_demux_deactivate_current_chain),
34194 (gst_ogg_demux_activate_chain):
34195 Extra debug output when activating/deactivating chains.
34196 * gst/playback/gstdecodebin.c: (gst_decode_bin_factory_filter),
34197 (is_demuxer_element), (try_to_link_1), (remove_element_chain),
34199 Remove a queue from our list when it becomes unlinked.
34200 Don't add queues to elements in class 'Demux' if they
34201 can only produce one pad
34203 2005-12-18 10:54:33 +0000 Julien Moutte <julien@moutte.net>
34205 gst-libs/gst/video/gstvideosink.c: Add a debug category.
34206 Original commit message from CVS:
34207 2005-12-18 Julien MOUTTE <julien@moutte.net>
34208 * gst-libs/gst/video/gstvideosink.c: (gst_video_sink_base_init),
34209 (gst_video_sink_get_type): Add a debug category.
34211 2005-12-18 00:56:07 +0000 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
34213 gst-libs/gst/rtp/gstbasertpdepayload.c: Handle downstream newsegment by sending our own newsegment before the next bu...
34214 Original commit message from CVS:
34215 2005-12-17 Philippe Khalaf <burger@speedy.org>
34216 * gst-libs/gst/rtp/gstbasertpdepayload.c:
34217 (gst_base_rtp_depayload_init), (gst_base_rtp_depayload_handle_sink_event):
34218 Handle downstream newsegment by sending our own newsegment before the
34219 next buffer to be released. (#323900)
34221 2005-12-18 00:41:10 +0000 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
34223 gst-libs/gst/rtp/gstbasertpdepayload.c: add queue delay to new segment as well (as opposed to just the first buffer)....
34224 Original commit message from CVS:
34225 2005-12-17 Philippe Khalaf <burger@speedy.org>
34226 * gst-libs/gst/rtp/gstbasertpdepayload.c:
34227 (gst_base_rtp_depayload_set_gst_timestamp):
34228 add queue delay to new segment as well (as opposed to just the first
34229 buffer). (bug #322347)
34231 2005-12-16 22:00:07 +0000 Stefan Kost <ensonic@users.sourceforge.net>
34233 ext/libvisual/visual.c: change some char* into char[]
34234 Original commit message from CVS:
34235 * ext/libvisual/visual.c: (make_valid_name):
34236 change some char* into char[]
34237 * gst/audiotestsrc/gstaudiotestsrc.c:
34238 (gst_audio_test_src_class_init), (gst_audio_test_src_do_seek),
34239 (gst_audio_test_src_create):
34240 * gst/audiotestsrc/gstaudiotestsrc.h:
34241 prepare to handle EOS and SEGMENT_DONE
34243 2005-12-16 12:32:37 +0000 Tim-Philipp Müller <tim@centricular.net>
34245 tests/check/generic/states.c: Blacklist cdparanoia element in state test.
34246 Original commit message from CVS:
34247 * tests/check/generic/states.c: (GST_START_TEST):
34248 Blacklist cdparanoia element in state test.
34250 2005-12-16 11:25:51 +0000 Benjamin Pineau <ben.pineau@gmail.com>
34252 gst/tcp/: Add <string.h> includes for memset and FD_ZERO (fixes #323878;
34253 Original commit message from CVS:
34254 * gst/tcp/gsttcp.c:
34255 * gst/tcp/gsttcpclientsink.c:
34256 * gst/tcp/gsttcpserversink.c:
34257 * gst/tcp/gsttcpserversrc.c:
34258 Add <string.h> includes for memset and FD_ZERO (fixes #323878;
34259 patch by: Benjamin Pineau).
34261 2005-12-15 14:43:38 +0000 Michael Smith <msmith@xiph.org>
34263 gst/videorate/gstvideorate.c: Fix timestamping for videorate when the first buffer it sees has a non-zero timestamp. ...
34264 Original commit message from CVS:
34265 * gst/videorate/gstvideorate.c: (gst_video_rate_blank_data),
34266 (gst_video_rate_chain):
34267 Fix timestamping for videorate when the first buffer it sees has a
34268 non-zero timestamp. Fix some misleading debug output.
34270 2005-12-15 10:30:14 +0000 Michael Smith <msmith@xiph.org>
34272 gst/audioresample/gstaudioresample.c: Don't leak all input buffers to audioresample.
34273 Original commit message from CVS:
34274 * gst/audioresample/gstaudioresample.c:
34275 Don't leak all input buffers to audioresample.
34277 2005-12-15 10:15:10 +0000 Tim-Philipp Müller <tim@centricular.net>
34279 ext/pango/gsttextoverlay.c: Don't operate on empty text buffers. Strip newlines and tabs only from the end of the tex...
34280 Original commit message from CVS:
34281 * ext/pango/gsttextoverlay.c: (gst_text_overlay_collected):
34282 Don't operate on empty text buffers. Strip newlines and
34283 tabs only from the end of the text, but leave them intact
34284 in the middle. Fix typo in gtk-doc description.
34286 2005-12-15 09:48:19 +0000 Tim-Philipp Müller <tim@centricular.net>
34288 gst/playback/: Make sure the video frame buffer we return to apps via the "frame" property always has caps set on it....
34289 Original commit message from CVS:
34290 * gst/playback/gstplaybasebin.c:
34291 * gst/playback/gstplaybin.c: (handoff):
34292 Make sure the video frame buffer we return to apps via the
34293 "frame" property always has caps set on it. Modify
34294 _gst_gvalue_set_object() macro to handle NULL objects
34297 2005-12-14 20:42:11 +0000 Stefan Kost <ensonic@users.sourceforge.net>
34299 gst/audiotestsrc/gstaudiotestsrc.*: Adjust to some recent api changes and add wtays new cool seeking capabillities
34300 Original commit message from CVS:
34301 * gst/audiotestsrc/gstaudiotestsrc.c:
34302 (gst_audio_test_src_class_init), (gst_audio_test_src_init),
34303 (gst_audio_test_src_setcaps), (gst_audio_test_src_src_query),
34304 (gst_audio_test_src_do_seek), (gst_audio_test_src_is_seekable),
34305 (gst_audio_test_src_create):
34306 * gst/audiotestsrc/gstaudiotestsrc.h:
34307 Adjust to some recent api changes and add wtays new cool seeking
34310 2005-12-14 17:58:48 +0000 Tim-Philipp Müller <tim@centricular.net>
34312 ext/alsa/: Helper functions to add device probing via the GstPropertyProbe interface to a class.
34313 Original commit message from CVS:
34314 * ext/alsa/Makefile.am:
34315 * ext/alsa/gstalsadeviceprobe.c:
34316 * ext/alsa/gstalsadeviceprobe.h:
34317 Helper functions to add device probing via the GstPropertyProbe
34318 interface to a class.
34319 * ext/alsa/gstalsamixer.h:
34320 Comment out GST_ALSA_MIXER, it returns a struct that's not
34322 * ext/alsa/gstalsamixer.c: (gst_alsa_mixer_open):
34323 Add some debug info.
34324 * ext/alsa/gstalsamixerelement.c:
34325 (gst_alsa_mixer_element_interface_supported),
34326 (gst_implements_interface_init),
34327 (gst_alsa_mixer_element_init_interfaces),
34328 (gst_alsa_mixer_element_class_init),
34329 (gst_alsa_mixer_element_finalize), (gst_alsa_mixer_element_init),
34330 (gst_alsa_mixer_element_set_property),
34331 (gst_alsa_mixer_element_get_property),
34332 (gst_alsa_mixer_element_change_state):
34333 * ext/alsa/gstalsamixerelement.h:
34334 Add 'device' and 'device-name' properties. Add GstPropertyProbe
34335 for device handling (gnome-volume-control will need that).
34337 2005-12-12 20:31:24 +0000 Christian Schaller <uraeus@gnome.org>
34341 * gst-plugins-base.spec.in:
34342 updates to activate cdparanoia plugin
34343 Original commit message from CVS:
34344 updates to activate cdparanoia plugin
34346 2005-12-12 19:13:09 +0000 Michael Smith <msmith@xiph.org>
34348 ext/ogg/gstoggdemux.c: Use the correct function to free list of typefind factories.
34349 Original commit message from CVS:
34350 * ext/ogg/gstoggdemux.c: (gst_ogg_type_find):
34351 Use the correct function to free list of typefind factories.
34353 2005-12-12 15:09:55 +0000 Wim Taymans <wim.taymans@gmail.com>
34355 gst/videotestsrc/gstvideotestsrc.*: Implement seeking in videotestsrc.
34356 Original commit message from CVS:
34357 * gst/videotestsrc/gstvideotestsrc.c:
34358 (gst_video_test_src_class_init), (gst_video_test_src_init),
34359 (gst_video_test_src_parse_caps), (gst_video_test_src_query),
34360 (gst_video_test_src_do_seek), (gst_video_test_src_is_seekable),
34361 (gst_video_test_src_create):
34362 * gst/videotestsrc/gstvideotestsrc.h:
34363 Implement seeking in videotestsrc.
34366 2005-12-12 15:06:46 +0000 Wim Taymans <wim.taymans@gmail.com>
34368 ext/cdparanoia/: Partially ported cdparanoia now that basesrc can support a plugin like this..
34369 Original commit message from CVS:
34370 * ext/cdparanoia/Makefile.am:
34371 * ext/cdparanoia/gstcdparanoia.c: (gst_paranoia_mode_get_type),
34372 (gst_paranoia_endian_get_type), (_do_init),
34373 (cdparanoia_class_init), (cdparanoia_init),
34374 (cdparanoia_set_property), (cdparanoia_get_property),
34375 (cdparanoia_do_seek), (cdparanoia_is_seekable),
34376 (cdparanoia_create), (cdparanoia_start), (cdparanoia_stop),
34377 (cdparanoia_convert), (cdparanoia_get_query_types),
34378 (cdparanoia_query), (cdparanoia_set_index),
34379 (cdparanoia_uri_set_uri):
34380 * ext/cdparanoia/gstcdparanoia.h:
34381 Partially ported cdparanoia now that basesrc can support a
34384 2005-12-12 13:03:50 +0000 Wim Taymans <wim.taymans@gmail.com>
34386 tests/examples/seek/scrubby.c: Set higher priority for bus events so they don't get reordered with gtk gui events.
34387 Original commit message from CVS:
34388 * tests/examples/seek/scrubby.c: (main):
34389 Set higher priority for bus events so they don't get reordered with
34391 * tests/examples/seek/seek.c: (do_seek), (start_seek), (stop_seek),
34392 (flush_toggle_cb), (main):
34393 Added checkbox do disable flushing seeks.
34394 Disable scrubbing when doing non flushing seeks.
34396 2005-12-12 09:52:37 +0000 Tim-Philipp Müller <tim@centricular.net>
34398 gst/subparse/gstsubparse.c: Implement some sort of event handling that doesn't rely on g_return_if_fail; make sure we...
34399 Original commit message from CVS:
34400 * gst/subparse/gstsubparse.c: (gst_sub_parse_init),
34401 (gst_sub_parse_do_seek), (gst_sub_parse_src_event), (parse_subrip),
34402 (parser_state_init), (handle_buffer), (gst_sub_parse_chain),
34403 (gst_sub_parse_sink_event), (gst_sub_parse_change_state):
34404 Implement some sort of event handling that doesn't rely on
34405 g_return_if_fail; make sure we always push the last chunk of an
34406 .srt out when we receive an EOS; use gst_pad_alloc_buffer; fix
34407 state change function; remove some old cruft. Seeking is still
34408 rather unlikely to work though.
34409 * tools/.cvsignore:
34412 2005-12-11 22:26:08 +0000 Julien Moutte <julien@moutte.net>
34414 sys/xvimage/xvimagesink.c: Fixed a leak of the current image reference when cleaning up.
34415 Original commit message from CVS:
34416 2005-12-11 Julien MOUTTE <julien@moutte.net>
34417 * sys/xvimage/xvimagesink.c: (gst_xvimagesink_change_state):
34418 Fixed a leak of the current image reference when cleaning up.
34419 Thanks to Arwed von Merkatz (alley_cat) for pointing it out.
34421 2005-12-09 10:23:42 +0000 Michael Smith <msmith@xiph.org>
34423 tools/: Remove gst-launch-ext. It doesn't work, and is no longer particularly useful.
34424 Original commit message from CVS:
34425 * tools/Makefile.am:
34426 * tools/gst-launch-ext-m.m:
34427 Remove gst-launch-ext. It doesn't work, and is no longer
34428 particularly useful.
34430 2005-12-08 18:53:57 +0000 Luca Ognibene <luogni@tin.it>
34432 ext/ogg/gstogmparse.c: don't pass random values to ogmparse convert function.
34433 Original commit message from CVS:
34434 * ext/ogg/gstogmparse.c: (gst_ogm_parse_sink_query):
34435 don't pass random values to ogmparse convert function.
34436 Make seeking possible in the exile1.ogm file.
34438 2005-12-07 18:51:35 +0000 Tim-Philipp Müller <tim@centricular.net>
34440 gst/playback/: Work around refcount problem with g_value_set_object() that occur if the core has been compiled agains...
34441 Original commit message from CVS:
34442 * gst/playback/gstplaybasebin.c: (gst_play_base_bin_get_property):
34443 * gst/playback/gstplaybin.c: (gst_play_bin_get_property):
34444 Work around refcount problem with g_value_set_object() that occur
34445 if the core has been compiled against GLib-2.6 (g_value_set_object()
34446 will only g_object_ref() the element, but the caller will
34447 gst_object_unref() it and bad things will happen due to the way
34448 GstObjects are refcounted in the GLib-2.6 case). Fixes problems with
34449 totem for people on FC4 using Thomas's 0.10 RPMs.
34451 2005-12-07 11:34:37 +0000 Edward Hervey <bilboed@bilboed.com>
34453 Time to welcome ogm to 0.10 :)
34454 Original commit message from CVS:
34455 Time to welcome ogm to 0.10 :)
34456 * ext/ogg/gstoggdemux.c: (internal_element_pad_added_cb),
34457 (gst_ogg_pad_typefind):
34458 Oggdemux can now properly typefind elements with dynamic pads.
34459 * ext/ogg/gstogmparse.c: (gst_ogm_parse_chain):
34460 Properly set caps on src pad, and set caps on outgoing buffers.
34462 2005-12-06 19:42:02 +0000 Thomas Vander Stichele <thomas@apestaart.org>
34465 * ext/alsa/gstalsamixer.h:
34466 * ext/alsa/gstalsamixerelement.h:
34467 * ext/alsa/gstalsamixeroptions.h:
34468 * ext/alsa/gstalsamixertrack.h:
34469 * ext/alsa/gstalsasink.c:
34470 * ext/alsa/gstalsasink.h:
34471 * ext/alsa/gstalsasrc.c:
34472 * ext/alsa/gstalsasrc.h:
34473 * ext/cdparanoia/gstcdparanoia.h:
34474 * ext/gnomevfs/gstgnomevfsuri.h:
34475 * ext/ogg/gstoggdemux.c:
34476 * ext/ogg/gstoggmux.c:
34477 * ext/pango/gsttextoverlay.h:
34478 * ext/theora/theoradec.c:
34479 * ext/theora/theoraenc.c:
34480 * ext/vorbis/vorbisdec.h:
34481 * ext/vorbis/vorbisenc.c:
34482 * ext/vorbis/vorbisenc.h:
34483 * ext/vorbis/vorbisparse.h:
34484 * gst-libs/gst/audio/gstaudioclock.h:
34485 * gst-libs/gst/audio/gstaudiosink.c:
34486 * gst-libs/gst/audio/gstaudiosink.h:
34487 * gst-libs/gst/audio/gstaudiosrc.c:
34488 * gst-libs/gst/audio/gstaudiosrc.h:
34489 * gst-libs/gst/audio/gstbaseaudiosink.c:
34490 * gst-libs/gst/audio/gstbaseaudiosink.h:
34491 * gst-libs/gst/audio/gstbaseaudiosrc.c:
34492 * gst-libs/gst/audio/gstbaseaudiosrc.h:
34493 * gst-libs/gst/audio/gstringbuffer.h:
34494 * gst-libs/gst/audio/multichannel.h:
34495 * gst-libs/gst/floatcast/floatcast.h:
34496 * gst-libs/gst/interfaces/colorbalance.c:
34497 * gst-libs/gst/interfaces/colorbalance.h:
34498 * gst-libs/gst/interfaces/colorbalancechannel.h:
34499 * gst-libs/gst/interfaces/mixer.h:
34500 * gst-libs/gst/interfaces/mixeroptions.h:
34501 * gst-libs/gst/interfaces/mixertrack.h:
34502 * gst-libs/gst/interfaces/navigation.h:
34503 * gst-libs/gst/interfaces/propertyprobe.h:
34504 * gst-libs/gst/interfaces/tuner.h:
34505 * gst-libs/gst/interfaces/tunerchannel.h:
34506 * gst-libs/gst/interfaces/tunernorm.h:
34507 * gst-libs/gst/interfaces/xoverlay.h:
34508 * gst-libs/gst/netbuffer/gstnetbuffer.h:
34509 * gst-libs/gst/riff/riff-ids.h:
34510 * gst-libs/gst/riff/riff-media.h:
34511 * gst-libs/gst/riff/riff-read.h:
34512 * gst-libs/gst/rtp/gstbasertpdepayload.h:
34513 * gst-libs/gst/rtp/gstbasertppayload.c:
34514 * gst-libs/gst/rtp/gstbasertppayload.h:
34515 * gst-libs/gst/rtp/gstrtpbuffer.c:
34516 * gst-libs/gst/rtp/gstrtpbuffer.h:
34517 * gst-libs/gst/tag/gsttageditingprivate.h:
34518 * gst-libs/gst/tag/gstvorbistag.c:
34519 * gst-libs/gst/tag/tag.h:
34520 * gst-libs/gst/video/video.h:
34521 * gst/adder/gstadder.c:
34522 * gst/adder/gstadder.h:
34523 * gst/audioconvert/audioconvert.c:
34524 * gst/audioconvert/audioconvert.h:
34525 * gst/audioconvert/gstaudioconvert.c:
34526 * gst/audioconvert/gstchannelmix.c:
34527 * gst/audioconvert/gstchannelmix.h:
34528 * gst/audiorate/gstaudiorate.c:
34529 * gst/audioresample/buffer.h:
34530 * gst/audioresample/functable.h:
34531 * gst/audioresample/gstaudioresample.c:
34532 * gst/audioresample/resample.h:
34533 * gst/ffmpegcolorspace/avcodec.h:
34534 * gst/ffmpegcolorspace/gstffmpegcodecmap.c:
34535 * gst/ffmpegcolorspace/gstffmpegcodecmap.h:
34536 * gst/ffmpegcolorspace/imgconvert.c:
34537 * gst/ffmpegcolorspace/imgconvert_template.h:
34538 * gst/playback/gstdecodebin.c:
34539 * gst/playback/gstplaybasebin.h:
34540 * gst/playback/gstplaybin.c:
34541 * gst/playback/gststreaminfo.h:
34542 * gst/tcp/gstfdset.c:
34543 * gst/tcp/gstfdset.h:
34544 * gst/tcp/gstmultifdsink.c:
34545 * gst/tcp/gstmultifdsink.h:
34546 * gst/tcp/gsttcp.h:
34547 * gst/tcp/gsttcpclientsrc.c:
34548 * gst/tcp/gsttcpclientsrc.h:
34549 * gst/tcp/gsttcpplugin.h:
34550 * gst/tcp/gsttcpserversink.c:
34551 * gst/tcp/gsttcpserversrc.c:
34552 * gst/typefind/gsttypefindfunctions.c:
34553 * gst/videorate/gstvideorate.c:
34554 * gst/videotestsrc/gstvideotestsrc.h:
34555 * gst/videotestsrc/videotestsrc.h:
34556 * sys/v4l/gstv4lcolorbalance.h:
34557 * sys/v4l/gstv4ltuner.h:
34558 * sys/v4l/gstv4lxoverlay.h:
34559 * sys/v4l/v4l_calls.h:
34560 * sys/v4l/videodev_mjpeg.h:
34561 * tests/check/elements/audioconvert.c:
34562 * tests/check/elements/audioresample.c:
34563 * tests/check/elements/audiotestsrc.c:
34564 * tests/check/elements/videotestsrc.c:
34565 * tests/check/elements/volume.c:
34566 * tests/examples/seek/scrubby.c:
34567 * tests/examples/seek/seek.c:
34569 Original commit message from CVS:
34572 2005-12-06 19:28:24 +0000 Thomas Vander Stichele <thomas@apestaart.org>
34574 * docs/libs/tmpl/gstaudio.sgml:
34575 * docs/libs/tmpl/gstcolorbalance.sgml:
34576 * docs/libs/tmpl/gstgconf.sgml:
34577 * docs/libs/tmpl/gstmixer.sgml:
34578 * docs/libs/tmpl/gstringbuffer.sgml:
34579 * docs/libs/tmpl/gsttuner.sgml:
34580 * docs/libs/tmpl/gstxoverlay.sgml:
34581 put back stability level
34582 Original commit message from CVS:
34583 put back stability level
34585 2005-12-05 18:11:49 +0000 Thomas Vander Stichele <thomas@apestaart.org>
34589 Original commit message from CVS:
34592 === release 0.10.0 ===
34594 2005-12-05 18:02:48 +0000 Thomas Vander Stichele <thomas@apestaart.org>
34600 * docs/libs/tmpl/gstcolorbalance.sgml:
34601 * docs/plugins/inspect/plugin-adder.xml:
34602 * docs/plugins/inspect/plugin-alsa.xml:
34603 * docs/plugins/inspect/plugin-audioconvert.xml:
34604 * docs/plugins/inspect/plugin-audiorate.xml:
34605 * docs/plugins/inspect/plugin-audioresample.xml:
34606 * docs/plugins/inspect/plugin-audiotestsrc.xml:
34607 * docs/plugins/inspect/plugin-decodebin.xml:
34608 * docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
34609 * docs/plugins/inspect/plugin-gnomevfs.xml:
34610 * docs/plugins/inspect/plugin-libvisual.xml:
34611 * docs/plugins/inspect/plugin-ogg.xml:
34612 * docs/plugins/inspect/plugin-pango.xml:
34613 * docs/plugins/inspect/plugin-playbin.xml:
34614 * docs/plugins/inspect/plugin-subparse.xml:
34615 * docs/plugins/inspect/plugin-tcp.xml:
34616 * docs/plugins/inspect/plugin-theora.xml:
34617 * docs/plugins/inspect/plugin-typefindfunctions.xml:
34618 * docs/plugins/inspect/plugin-video4linux.xml:
34619 * docs/plugins/inspect/plugin-videorate.xml:
34620 * docs/plugins/inspect/plugin-videoscale.xml:
34621 * docs/plugins/inspect/plugin-videotestsrc.xml:
34622 * docs/plugins/inspect/plugin-volume.xml:
34623 * docs/plugins/inspect/plugin-vorbis.xml:
34624 * docs/plugins/inspect/plugin-ximagesink.xml:
34625 * docs/plugins/inspect/plugin-xvimagesink.xml:
34627 Original commit message from CVS: